[Asterisk-Users] SIP - Native Bridging - sipgate.de
Hi,
I have the following two tries. While the first calls out just fine, the
second call does not work.
My asterisk box is behind a NATed fli4l fw. Port 1-2, 5060, 5004
are all forwarded to the asterisk box.
What throws me off, is the fact, that the first call worked fine, and
the second did not.
extension.conf
==
exten = _*9.,1,Dial,SIP/${EXTEN:[EMAIL PROTECTED],30,tr
sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
nat = yes
canreinvite=no
register = 4318776:[EMAIL PROTECTED]/4318776
[sipgate]
secret=xxx
username=4318776
fromuser=4318776
fromdomain=sipgate.net
type=friend
host=sipgate.de
nat=yes
dtmfmode=rfc2833
canreinvite=no
context=in-sipgate
Any and all help is greatly apprciated.
-- Executing Dial(SIP/25-ebe3, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
-- SIP/sipgate-db58 is making progress passing it to SIP/25-ebe3
== Spawn extension (default, *901729731418, 1) exited non-zero on
'SIP/25-ebe3'
-- Executing Dial(SIP/25-e2f9, SIP/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
-- SIP/sipgate-e2ee is making progress passing it to SIP/25-e2f9
-- SIP/sipgate-e2ee answered SIP/25-e2f9
-- Attempting native bridge of SIP/25-e2f9 and SIP/sipgate-e2ee
== Spawn extension (default, *901715828552, 1) exited non-zero on
'SIP/25-e2f9
--
Best regards
Peer Oliver Schmidt
the internet company
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Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
A comment and a question about the latest version:
First, the question: What is the organizing principle of the Asterisk
Minutes chart, with respect to the ordering of the various days? It
seems random. . . I also assume the last column is the average number of
minutes per call?
As to the fix, I had to make a change to ~/lib/Class.Table.php in order
to pass syntactic muster with Postgres 7.4.1, the corrected version of
which is:
if (DB_TYPE == postgres){
// $sql_limit = LIMIT $limite;
$sql_limit = LIMIT $limite OFFSET
$current_record;
}else{
$sql_limit = LIMIT
$current_record,$limite;
}
}
Otherwise it now works beautifully for me (Apache 2.0.49, PHP 5.0.0RC1,
Postgres 7.4.1)
Thanks.
B.
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Re: [Asterisk-Users] SIP - Native Bridging - sipgate.de - Additional information
I hate to answer myself, but another thing: The first call worked fine. This is not true. The moment the other side answers the call, Asterisk -- Attempting native bridge of SIP/25-e2f9 and SIP/sipgate-e2ee This obviously can't work. How can I make sure asterisk never tries to do this?! canreinvite is set to no TIA -- Best regards Peer Oliver Schmidt the internet company ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 and SCSI
I had a similar issue when installing my G.729 licences. I contacted Digium support and an engineer logged into my system and performed some hocus pocus and got it working for me ... -Original Message- From: Derek Samford [mailto:[EMAIL PROTECTED] Sent: 25 March 2004 18:29 To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] G.729 and SCSI If memory servers, and everyone feel free to flame away if it serves badly, the library only searches hda,hdb,hdc, and hdd. Try switching where your controller is, that may solve it. Derek -Original Message- From: Sergio Serrano [mailto:[EMAIL PROTECTED] Sent: Thursday, March 25, 2004 12:17 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] G.729 and SCSI Yes I have mounted CDROM first with automount(/dev/cdrom) and second manually(/dev/hde) but nothing. Any idea? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Andrew Thompson Enviado el: jueves, 25 de marzo de 2004 17:59 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] G.729 and SCSI Sergio Serrano wrote: Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio Here is the wiki page for g729: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing It's not specifically listed there, but the licensing process has issues with SCSI only systems. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreeBSD Segmentation Fault on start up
Currently the asterisk port is blocked due to vulnerabilities in pwlib. Chris - Original Message - From: Joe Lewis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 9:30 PM Subject: [Asterisk-Users] FreeBSD Segmentation Fault on start up To all; I've got two installations of asterisk. The last one (installed a few days ago) is from the FreeBSD ports, and many thanks, because it compiled BEAUTIFULLY! However, I can't run it. Everytime I start asterisk, I get a segmentation fault. asterisk -c reveals : [...snip...] [codec_gsm.so] = (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 1 == Registered translator 'lintogsm' from format SLINR to GSM, cost 5 [codec_mp3_d.so] = (MP3/PCM16 (signed linear) Translator (Decoder only)) Segmentation fault (core dumped) So, I check the core dump to see what I can find, and get : Reading symbols from /usr/local/lib/asterisk/modules/codec_gsm.so... (no debugging symbols found)...done. Loaded symbols for /usr/local/lib/asterisk/modules/codec_gsm.so Reading symbols from /usr/local/lib/asterisk/modules/codec_mp3_d.so... (no debugging symbols found)...done. Loaded symbols for /usr/local/lib/asterisk/modules/codec_mp3_d.so Reading symbols from /libexec/ld-elf.so.1...(no debugging symbols found)... done. Loaded symbols for /libexec/ld-elf.so.1 #0 0x2953ff53 in unpack_huff () from /usr/local/lib/asterisk/modules/codec_mp3_d.so (gdb) Would there, by chance, be a missing library or package that I need? Could someone point out a possible solution? (Maybe the port assumed I have an mp3 library installed?) Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple IAX register lines?
Hi, Having a small problem here and wondering if anyone else has seen it.. My Asterisk box is behind NAT so I need to register with the external IAX Asterisk boxes for calls to be received.. Up till yesterday I only needed to register with a single external IAX server and all was working fine.. Now I need to register with a second external IAX server.. So I now have two register lines in my IAX.conf.. The problem is that Asterisk only uses the first one and is ignoring the second.. If I comment out the first one then asterisk will happily use the second one but it does not seem to be happy using both at the same time.. Is anyone else having this problem? is it a bug? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with SIPPS and ilbc
Well the warning is: Mar 18 16:46:47 WARNING[737296]: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (38)? And the sound is cripple (really broken). Any solution? -- Alejandro Escanero Blanco Administrador Sistemas CEC. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX drops calls exactly 5 secs into the call
Hi List, Two boxes A has a PRI B terminates SIP devices A --IAX-- B Both on the same switch, same IP network. Call from PSTN to A gets pushed via IAX to B - Sip device with no problems. Call from Sip device - B via IAX - A - PSTN will drop exactly 5 seconds after the call is answered. I've built with 0.7.2, 1_0_Stable develetc Any clue / hints ?? This strongly resembles a SIP problem I had with Grandstream SIP phones. The phones did not like the codec and refused to ACK. The Invite protocol should go; Phone sends INVITE Asterisk sends 200 OK with preferred CODEC Phone refuses to ACK (does not like CODEC) Media exchange begins (call seems OK) A few seconds later * dumps call cos no ACK Do a dump of the protocol (Ethereal, tcpdump) and check it out. Might be a similar problem. Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New soundfiles from Allison posted
John Todd wrote: I've finally uploaded the newest (LARGE) list of sound clips in .gsm format to the bugtracker. Great thanks a lot :) Assume they'll make it into CVS tarball sometime soonish. Please see http://bugs.digium.com/bug_view_page.php?bug_id=985 for details and a full sound file list (and a tarball of the sounds in gsm format.) So, should we open up a new bug for the next set of requests? ;) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX register lines?
Works like a charm for me. I have both VoicePulse and NuPhone registered in IAX. Depending upon the phone nr dialed, I send a call via NP or VP. And yes, my [*] box is behind a NAT. Include the relevant lines of your iax.conf so we can take a look. Cheers, Willy - Original Message Follows - Hi, Having a small problem here and wondering if anyone else has seen it.. My Asterisk box is behind NAT so I need to register with the external IAX Asterisk boxes for calls to be received.. Up till yesterday I only needed to register with a single external IAX server and all was working fine.. Now I need to register with a second external IAX server.. So I now have two register lines in my IAX.conf.. The problem is that Asterisk only uses the first one and is ignoring the second.. If I comment out the first one then asterisk will happily use the second one but it does not seem to be happy using both at the same time.. Is anyone else having this problem? is it a bug? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Several H323 bugfixes - working SIP - H.323 translator
this patch however worked for me, all calls through the patched
chan_h323 are ok, hold, transfer, etc works perfectly. except that there
is no music on hold, while in fact asterisk shows that it is playing,
yet there is no audio heard on the callmanager side.
so i had test on both oh323 0.5.10 and h323(patched) cvs march 20 the
problem with oh323 is that when a call is placed on hold by a
callmanager phone, after resuming, the audio from * to ccm is lagged by
3-4 seconds. while the audio from ccm to * is ok. i already posted this
problem in version 0.5.5. has anybody found a workaround for this?
On Fri, 2004-03-19 at 18:25, Paul Cheng wrote:
Hi,
The patches also did not help us and in fact created some new problems.
The old chan_h323 could pass on early audio and provider messages, but
after the patch, this capability is gone and the channel only rings and
rings while the provider is sending the message.
We've had no problems with the existing chan_h323 other than that it
doesn't return the right indication state to Asterisk, so Asterisk
can't branch for busy versus congestion.
But this is obviously only for our setup.
On Mar 19, 2004, at 9:12 AM, Marian Durkovic wrote:
On Thu, Mar 18, 2004 at 12:22:57PM -0500, Billy Huddleston wrote:
I just tried this, and it's not working for me.. I can't call a 2600
or a
CCM... What version of OpenH323 and PWLIB did you all use?
Are you able to call those without the patches? If not, the patches
won't
help you, since you probably have some other problem..
M.
- Original Message -
From: Marian Durkovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 18, 2004 10:35 AM
Subject: [Asterisk-Users] Several H323 bugfixes - working SIP -
H.323
translator
Hi all,
in an effort to create a SIP - H.323 translator we've found and
fixed
several problems in H.323 channel. These inlcude:
for SIP-H.323 calls
- no ringback tone
- ringback not related to H.323 events
- one-way audio with Cisco CallManager
- incorrect Caller ID
for H.323-SIP calls
- not able to establish call with Cisco IOS 12.3(4)T
- ringback not related to SIP events
- no support for 183 Call Progress
- incorrect Caller ID
Please find the patches against aterisk 0.7.2 release below.
M.
-
-
Marian Durkovic network manager
Slovak Technical University Tel: +421 2 524 51 301
Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED]
-
-
---
---
Marian Durkovic network manager
Slovak Technical University Tel: +421 2 524 51 301
Computer Centre, Nam. Slobody 17 Fax: +421 2 524 94 351
812 43 Bratislava, Slovak RepublicE-mail/sip: [EMAIL PROTECTED]
---
---
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Re: [Asterisk-Users] Multiple IAX register lines?
[EMAIL PROTECTED] wrote: Works like a charm for me. I have both VoicePulse and NuPhone registered in IAX. Depending upon the phone nr dialed, I send a call via NP or VP. And yes, my [*] box is behind a NAT. Include the relevant lines of your iax.conf so we can take a look. Cheers, Willy There is nothing to it really register lines are pretty simple.. register = user1:[EMAIL PROTECTED] register = user2:[EMAIL PROTECTED] From the cli iax2 show registry only shows the first entry.. These are for inbound services not outbound, I didn't think it was nesesary to register for outbound calls because the call is being initiated from inside.. Later.. - Original Message Follows - Hi, Having a small problem here and wondering if anyone else has seen it.. My Asterisk box is behind NAT so I need to register with the external IAX Asterisk boxes for calls to be received.. Up till yesterday I only needed to register with a single external IAX server and all was working fine.. Now I need to register with a second external IAX server.. So I now have two register lines in my IAX.conf.. The problem is that Asterisk only uses the first one and is ignoring the second.. If I comment out the first one then asterisk will happily use the second one but it does not seem to be happy using both at the same time.. Is anyone else having this problem? is it a bug? Thanks.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX register lines?
Gus,
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED]
register = user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
nesesary to register for outbound calls because the call is being
initiated from inside..
If the iax destination site is static, you don't even need the registration
process. Simply set up a type=user (for inbound iax calls) with some
appropriate security parameters (host, secret, context). The associated
extensions.conf entry might look something like:
exten = _9.,1,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN:1})
which will connect with type=user just fine. Have several of these in
production.
Of coarse, if you don't have access/cooperation to the remote iax2 servers,
or if any site is using dynamic IP addresses, then your stuck with
resolving the original registration problem. (Does a iax2 debug show
anything useful?)
Rich
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[Asterisk-Users] SIP Call Progress
Hi, I am a newbie and am currently writing an application for making outbound calls for a reminder service.I am using AGI for that Problem isfor SIP calls. As soon as the call goes through (even ringing), Asterisk says that the call is answered. Checking CHANNEL STATUS gives me 6 even though the line is ringing. For Zap calls, there is no problem. It detects answered correctly. I am using callprogress=yes, in zapata.conf. I am placing SIP calls through sipphone iconnectthere. Can someone point me to the right direction? Thanx in advance
Re: [Asterisk-Users] Multiple IAX register lines?
Rich Adamson wrote:
Gus,
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED]
register = user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
nesesary to register for outbound calls because the call is being
initiated from inside..
If the iax destination site is static, you don't even need the registration
process. Simply set up a type=user (for inbound iax calls) with some
appropriate security parameters (host, secret, context). The associated
extensions.conf entry might look something like:
exten = _9.,1,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN:1})
which will connect with type=user just fine. Have several of these in
production.
Of coarse, if you don't have access/cooperation to the remote iax2 servers,
or if any site is using dynamic IP addresses, then your stuck with
resolving the original registration problem. (Does a iax2 debug show
anything useful?)
Rich
Thats the problem I have a dynamic IP on my side which is why I need the
register line in the iax.conf..
iax2 debug shows that it is registering the first line but not the second..
Later..
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[Asterisk-Users] Help needed (New to Asterisk)
Hello friends, Just yesterday i joined asterisk mailing list. We want to use it in our company. Can someone tell me which version or type of Linux would best work for it and also what should be the configuration of machine (hardware configuration) to install Asterisk? That would help me a lot.ThanksYash.Join Excite! - http://www.excite.comThe most personalized portal on the Web!
Re: [Asterisk-Users] SoftFAX/spandsp
Eric Wieling wrote: On Thu, 2004-03-25 at 09:33, Steve Underwood wrote: exten = 5678,1,txfax(/tmp/testfax.tif|caller) There are a zillion fax and tiff formats. I'm trying to figure out what output format I should tell GhostScript to use. Any suggestions on which format to try? These are the formats GhostScript can output: faxg3 faxg32d faxg4 tiff12nc tiff24nc tiffcrle tiffg3 tiffg32d tiffg4 tifflzw tiffpack These are not TIFF: faxg3 faxg32d faxg4 These are TIFF, but not for faxes: tiff12nc tiff24nc tiffcrle tifflzw tiffpack These are TIFF formats for faxes: tiffg3 tiffg32d tiffg4 tiffg3 is the commonest for used for fax work. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Semi OT: WiSIP and WEP
On Thu, 25 Mar 2004 23:29:52 -0600, Christian Hoffmeyer wrote: - Original Message - From: Clif Jones [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 8:06 PM Subject: Re: [Asterisk-Users] Semi OT: WiSIP and WEP Are you saying that the clicking only occurs when 128-bit WEP is enabled? Verified. The WiSip phone has almost perfect quality with no WEP encryption, near perfect with 64 bit encryption, and clicks approximately 2-3 times per second with 128 bit encryption. Christian Hoffmeyer YottaDot Solutions Huntsville, AL So, let's recap...that would make the phone about useless unless you want to run your WLAN totally wide open unencrypted. Doesn't make any sense in an office or even SOHO environment. That's not indicative of a manufacturer in touch with their intended market. I guess I'll have to wait for a wifi phone that supports WPA on 802.11g. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] Of course, I'm equal parts cynicism and apathy...I'm always willing to beleive the worst if it doesn't take too much effort. - Denis Miller ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: SoftFAX/spandsp
Hi Reynaldo, The is the only report of seg faults I have had with recent versions of spandsp. There was an older version (spandsp-0.0.1b I think) which had a silly bug that caused seg faults. Some people have had older versions of libtiff installed, which seem to cause seg faults. If you are use RH9 that should not be the case. It seems version libtiff 3.6.1 is buggy, and should not be used for fax work. It seems it fails with Hylafax as well as with spandsp. Since the segmentation fault occurs as processing of the image data for the first page starts, something about libtiff still seems like a possibility. Regards, Steve Reynaldo Simbulan wrote: Hi, I've been testing the soft fax but I am getting this segmentation fault whenever I receive a fax. Can somebody help me please? I've loaded asterisk from CVS and running Redhat 9.0 and downloaded the latest spandsp. I am sending a fax from Windows XP fax software thru a ringmaster then to X100P. WindowsXP -- modem --- ringmaster (CO simulator) --- X100P --- * == Spawn extension (prepaid, fax, 0) exited non-zero on 'Zap/1-1' -- Executing RxFAX(Zap/1-1, /root/test.tiff) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 78 61 46 TSI without final frame tag Remote fax gave TSI as: Fax DCS: 83 00 c6 70 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Coarse carrier frequency 1683.64 (18) Fast carrier down Trainability test failed - longest run of zeros test was 0 FTT: 44 Fast carrier up Coarse carrier frequency 1699.71 (64) Training error 9.552597 Training succeeded (constellation mismatch 11.776280) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page Segmentation fault ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Drop / Call Tranfer - tranfering a call to a different number.
There is a calling card AGI script that you can probably find in the archives by searching for calling card agi. Also I posted a good little script for unsupervised call transfer of incoming calls some time ago. I'm sure people have improved it since then. The only thing you need is a centrex style line for us its called local link and most standard business lines have the required feature available. Basically the agi script will handle your authentication routine and then prompt for the number you want to dial. Once this is done asterisk will flash the line, wait a couple seconds for the line to switch over senddtmf out which will effectively dial the number flash again to link the calls and then hangup effectively freeing up that line. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 7:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Drop / Call Tranfer - tranfering a call to a different number. First, I am very new to this software. If I missed a searchable archive, please point me in the right direction. I am wishing to know if Asterisk can be used to do a Call Drop scenario. This is where someone calls, Asterisk answers, ask for the number that the person wishes to dial, gets the PIN, and then completes the call to the number they desired. Once the connection is completed, this software/service is no longer in the call loop. Typically this scenario is used to offer a wider calling area. Called Metro or Extended Metro in our area. There are many people in area that this feature is not available from their phone company, or they don't want to pay much for it, as they don't make many calls. I am certainly willing to provide more information, but I wanted to find out if Asterisk was even something that could do it- or be modified to do so. Thanks, John Chapman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX register lines?
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED]
register = user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
nesesary to register for outbound calls because the call is being
initiated from inside..
If the iax destination site is static, you don't even need the registration
process. Simply set up a type=user (for inbound iax calls) with some
appropriate security parameters (host, secret, context). The associated
extensions.conf entry might look something like:
exten = _9.,1,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN:1})
which will connect with type=user just fine. Have several of these in
production.
Of coarse, if you don't have access/cooperation to the remote iax2 servers,
or if any site is using dynamic IP addresses, then your stuck with
resolving the original registration problem. (Does a iax2 debug show
anything useful?)
Rich
Thats the problem I have a dynamic IP on my side which is why I need the
register line in the iax.conf..
iax2 debug shows that it is registering the first line but not the second..
I assume you've tried the easy stuff... separate the two statements with
some additional cf-lf, add a useless/redundant statement in between the two
registrations, move one of them to the bottom of the config file,
database show to ensure entries are accurate, etc, etc.
(I had a similar problem months ago that involved an incorrect /IAX entry
in the database. If I recall correctly, iaxtel changed IP addresses or
something like that. At that time, I didn't know some of these things were
actually stored in the database, and no one responded on the list to my
question. If I would have simply deleted the database entry (assuming I
knew it was there in the first place), my problem would have been fixed.)
Rich
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[Asterisk-Users] Watchguard Firebox 1000 and Asterisk
Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I have asterisk on public side and phones on the private side. I am able to get the phones to register and make outbound calls but the inbound calls are intermittent. I have NAT enable in asterisk and on the Cisco 7960. Any insight would be appreciated. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX register lines?
Rich Adamson wrote: Thats the problem I have a dynamic IP on my side which is why I need the register line in the iax.conf.. iax2 debug shows that it is registering the first line but not the second.. I assume you've tried the easy stuff... separate the two statements with some additional cf-lf, add a useless/redundant statement in between the two registrations, move one of them to the bottom of the config file, database show to ensure entries are accurate, etc, etc. Yup done all that.. makes no difference.. I have just built from the latest CVS as well and its still not registering to the two servers.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Watchguard Firebox 1000 and Asterisk
Has any had any experiences with Watchguard Firebox 1000 and Asterisk. I have asterisk on public side and phones on the private side. I am able to get the phones to register and make outbound calls but the inbound calls are intermittent. I have NAT enable in asterisk and on the Cisco 7960. I don't believe the Watchguard products are sip aware, therefore you will need to address all of the nat'ing issues common to running sip and rtp through the box. You are likely to have to change the registration frequency on the C7960 to a shorter period of time as I'd bet the Watchguard will timeout the nat table entries sooner then the phone system. A packet sniffer (eg, ethereal) will be your friend towards resolving the problem. Without some indication as to exactly which udp ports are being used for rtp, etc, there isn't going to be much help from the list. I can tell you that I had a snom 200 working through a watchguard in a very similar setup a couple of months ago. I did not have to make any changes to the watchguard in that case at all. (But, the watchguard was at a school where outbound traffic was basically unrestricted. Sound was choppy, but they had a well-known overloaded firewall too.) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Semi OT: WiSIP and WEP
That sounds about right. I sometimes use my PDA with xlite installed on it, and I get clicks when I am using 128 Bit wep. I now have 2 access points. One uses 256 Bit WEP and is only for my .g data network, and then I have a Orinoco rg-1000 in access point mode and the provides access for my pda. The second is very secure, it will only allow communication from registered devices. Have a good one! Zac Zac Amsler, Technical Team WNOC.COM Wireless Network Operations Center Phone: (801) 606-8047 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, March 26, 2004 7:41 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Semi OT: WiSIP and WEP On Thu, 25 Mar 2004 23:29:52 -0600, Christian Hoffmeyer wrote: - Original Message - From: Clif Jones [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 8:06 PM Subject: Re: [Asterisk-Users] Semi OT: WiSIP and WEP Are you saying that the clicking only occurs when 128-bit WEP is enabled? Verified. The WiSip phone has almost perfect quality with no WEP encryption, near perfect with 64 bit encryption, and clicks approximately 2-3 times per second with 128 bit encryption. Christian Hoffmeyer YottaDot Solutions Huntsville, AL So, let's recap...that would make the phone about useless unless you want to run your WLAN totally wide open unencrypted. Doesn't make any sense in an office or even SOHO environment. That's not indicative of a manufacturer in touch with their intended market. I guess I'll have to wait for a wifi phone that supports WPA on 802.11g. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] Of course, I'm equal parts cynicism and apathy...I'm always willing to beleive the worst if it doesn't take too much effort. - Denis Miller ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help needed (New to Asterisk)
Yash, Just yesterday i joined asterisk mailing list. We want to use it in our company. Can someone tell me which version or type of Linux would best work for it and also what should be the configuration of machine (hardware configuration) to install Asterisk? I did it on Debian 3.0 with kernel 2.4.24 and on Redhat 9.0. PC's are 1GHz 512MB RAM. The System is slightly loaded Jakob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Semi OT: WiSIP and WEP
Ugh, I didn't have a chance to try it last night. I guess I'll have to pull out my old Apple 802.11b AP and use that for the phone and/or non-secure connections from friend's laptops, etc. Good real world experience, is there a way to feed this back to Jeff, maybe for a WiSIP v2? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Zac Amsler Sent: Friday, March 26, 2004 9:47 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Semi OT: WiSIP and WEP That sounds about right. I sometimes use my PDA with xlite installed on it, and I get clicks when I am using 128 Bit wep. I now have 2 access points. One uses 256 Bit WEP and is only for my .g data network, and then I have a Orinoco rg-1000 in access point mode and the provides access for my pda. The second is very secure, it will only allow communication from registered devices. Have a good one! Zac Zac Amsler, Technical Team WNOC.COM Wireless Network Operations Center Phone: (801) 606-8047 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Friday, March 26, 2004 7:41 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Semi OT: WiSIP and WEP On Thu, 25 Mar 2004 23:29:52 -0600, Christian Hoffmeyer wrote: - Original Message - From: Clif Jones [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 8:06 PM Subject: Re: [Asterisk-Users] Semi OT: WiSIP and WEP Are you saying that the clicking only occurs when 128-bit WEP is enabled? Verified. The WiSip phone has almost perfect quality with no WEP encryption, near perfect with 64 bit encryption, and clicks approximately 2-3 times per second with 128 bit encryption. Christian Hoffmeyer YottaDot Solutions Huntsville, AL So, let's recap...that would make the phone about useless unless you want to run your WLAN totally wide open unencrypted. Doesn't make any sense in an office or even SOHO environment. That's not indicative of a manufacturer in touch with their intended market. I guess I'll have to wait for a wifi phone that supports WPA on 802.11g. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] Of course, I'm equal parts cynicism and apathy...I'm always willing to beleive the worst if it doesn't take too much effort. - Denis Miller ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature
[Asterisk-Users] Newbie Softphone Problem
Hi all, Sorry to bother you all with this, but I'm sure I've done something stupid. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+phone+sjphone (substituting 100 with 377 - the extention I was for this softphone) but when I try to register the softphone client I get the following on sip debug: Sip read: REGISTER sip:10.216.6.50 SIP/2.0 Content-Length: 0 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=91157703 CSeq: 2 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 10.216.6.127:5060 8 headers, 0 lines Using latest request as basis request Sending to 10.216.6.127 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.216.6.127:5060 From: sip:[EMAIL PROTECTED];tag=91157703 To: sip:[EMAIL PROTECTED];tag=as1b2f3686 Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 10.216.6.127:5060 Mar 26 14:58:32 NOTICE[-1179178064]: chan_sip.c:5568 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '10.216.6.127' Urgent handler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Multiple IAX register lines?
In you iax.conf file, are you using type=friend? I seem to remember a
discussion about problems if using type=friend instead of type=user for
multiple registered servers with the same userid. I may have the details
messed up on this though!
I have the same setup your describing and it works great!
In the iax.conf file I have broken the inbound and outbound into 2 separate
stanzas, i.e. -
; for inbound from Nufone
[NuFone]
Type=user
; for outbound to Nufone
[NuFone-peer]
Type=peer
Hope this helps.
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Friday, March 26, 2004 5:28 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Multiple IAX register lines?
Rich Adamson wrote:
Gus,
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED] register =
user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
nesesary to register for outbound calls because the call is being
initiated from inside..
If the iax destination site is static, you don't even need the
registration process. Simply set up a type=user (for inbound iax calls)
with some appropriate security parameters (host, secret, context). The
associated extensions.conf entry might look something like:
exten = _9.,1,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN:1})
which will connect with type=user just fine. Have several of these in
production.
Of coarse, if you don't have access/cooperation to the remote iax2
servers, or if any site is using dynamic IP addresses, then your stuck
with resolving the original registration problem. (Does a iax2 debug
show anything useful?)
Rich
Thats the problem I have a dynamic IP on my side which is why I need the
register line in the iax.conf..
iax2 debug shows that it is registering the first line but not the second..
Later..
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Re: [Asterisk-Users] Help needed (New to Asterisk)
Jakob Strebel wrote: Yash, Just yesterday i joined asterisk mailing list. We want to use it in our company. Can someone tell me which version or type of Linux would best work for it and also what should be the configuration of machine (hardware configuration) to install Asterisk? I did it on Debian 3.0 with kernel 2.4.24 and on Redhat 9.0. PC's are 1GHz 512MB RAM. The System is slightly loaded There's a lot of information about systems and hardware for Asterisk on http://www.voip-info.org Please visit there for information on how to start and to read the FAQ! Welcome to the Asterisk community. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple IAX register lines?
Hi,
I have just solved it, I deleted the iax.conf and created a new one
(exactly the same) and now its working.. I guess there was a gremlin in
the text file somewhere the was twisting things up..
Thanks to everyone for you help and suggestions..
later..
Ed Rubright wrote:
In you iax.conf file, are you using type=friend? I seem to remember a
discussion about problems if using type=friend instead of type=user for
multiple registered servers with the same userid. I may have the details
messed up on this though!
I have the same setup your describing and it works great!
In the iax.conf file I have broken the inbound and outbound into 2 separate
stanzas, i.e. -
; for inbound from Nufone
[NuFone]
Type=user
; for outbound to Nufone
[NuFone-peer]
Type=peer
Hope this helps.
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: Friday, March 26, 2004 5:28 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Multiple IAX register lines?
Rich Adamson wrote:
Gus,
There is nothing to it really register lines are pretty simple..
register = user1:[EMAIL PROTECTED] register =
user2:[EMAIL PROTECTED]
From the cli iax2 show registry only shows the first entry..
These are for inbound services not outbound, I didn't think it was
nesesary to register for outbound calls because the call is being
initiated from inside..
If the iax destination site is static, you don't even need the
registration process. Simply set up a type=user (for inbound iax calls)
with some appropriate security parameters (host, secret, context). The
associated extensions.conf entry might look something like:
exten = _9.,1,Dial(IAX2/userid:[EMAIL PROTECTED]/${EXTEN:1})
which will connect with type=user just fine. Have several of these in
production.
Of coarse, if you don't have access/cooperation to the remote iax2
servers, or if any site is using dynamic IP addresses, then your stuck
with resolving the original registration problem. (Does a iax2 debug
show anything useful?)
Rich
Thats the problem I have a dynamic IP on my side which is why I need the
register line in the iax.conf..
iax2 debug shows that it is registering the first line but not the second..
Later..
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[Asterisk-Users] isdn30e E100P configuration
I have an 8 channel isdn30e coming in from BT. Can anyone point me to sample zap*.conf that will work. Thanks JC
Re: [Asterisk-Users] Semi OT: WiSIP and WEP
- Original Message - From: Adams, Gavin [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 26, 2004 9:01 AM Subject: RE: [Asterisk-Users] Semi OT: WiSIP and WEP Good real world experience, is there a way to feed this back to Jeff, maybe for a WiSIP v2? I do not believe that PulverInnovations designs these phones. Christian Hoffmeyer YottaDot Solutions Huntsville, AL (w) 256.859.4508 (c)256.655.0321 (iax) 700.859.4508 Ask me about Asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie Softphone Problem
All, I figured it out Typo in the secret field... How embarrassing... Matt -Original Message- From: Matt Bridges [mailto:[EMAIL PROTECTED] Sent: 26 March 2004 15:04 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie Softphone Problem Hi all, Sorry to bother you all with this, but I'm sure I've done something stupid. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+phone+sjphone (substituting 100 with 377 - the extention I was for this softphone) but when I try to register the softphone client I get the following on sip debug: Sip read: REGISTER sip:10.216.6.50 SIP/2.0 Content-Length: 0 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] From: sip:[EMAIL PROTECTED];tag=91157703 CSeq: 2 REGISTER To: sip:[EMAIL PROTECTED] Via: SIP/2.0/UDP 10.216.6.127:5060 8 headers, 0 lines Using latest request as basis request Sending to 10.216.6.127 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.216.6.127:5060 From: sip:[EMAIL PROTECTED];tag=91157703 To: sip:[EMAIL PROTECTED];tag=as1b2f3686 Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 10.216.6.127:5060 Mar 26 14:58:32 NOTICE[-1179178064]: chan_sip.c:5568 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '10.216.6.127' Urgent handler ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreeBSD Segmentation Fault on start up
On Thursday 25 March 2004 15:30, Joe Lewis wrote: I've got two installations of asterisk. The last one (installed a few days ago) is from the FreeBSD ports, and many thanks, because it compiled BEAUTIFULLY! However, I can't run it. Everytime I start asterisk, I get a segmentation fault. Please email the ports maintainer and ask for a fix. The Asterisk community is by-and-large not responsible for broken FreeBSD ports. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: 0.7.2 with cisco router 7960
Our cisco router have these dial peers: dial-peer voice 900 pots application session destination-pattern 5000 port 1/0/0 ! dial-peer voice 800 pots application session destination-pattern 9 port 1/1/1 ! dial-peer voice 701 pots application session destination-pattern 3003 port 1/0/1 ! dial-peer voice 10 pots application session destination-pattern 13T port 0/0:1-- Channelized E1 ! dial-peer voice 5 pots incoming called-number X00 direct-inward-dial ! dial-peer voice 35 pots application session destination-pattern 12T port 1/1/1 ! dial-peer voice 36 pots application session destination-pattern 14T port 1/1/0 ! dial-peer voice 1 voip application session destination-pattern ... session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 4 pots incoming called-number X10 direct-inward-dial ! dial-peer voice 6 pots incoming called-number X11 direct-inward-dial ! dial-peer voice 7 pots incoming called-number X12 direct-inward-dial ! = When the call is from PSTN, detection of DTMF by Asterisk+Cisco 2600 works pretty well; but when the call is from Cisco 7960 phone thru ASTERISK+Cisco 2600 to PSTN (like IVR o PBX) always DTMF tones (for long number example 4 or more) aren´t recognized or it has wrong detection (I digit 9228373 but PBX in PSTN seen 928373 or 9287 or 922283). am I missing anything? Regards Daniel Pd. What is meaning of CME? Kurt Pasewaldt writes: What does your VoIP dial peer look like? Does it have dtmf-relay rtp-nte under the VoIP dial peer. This will enable RC2833. This assume you are not running the CME load on the router. Kurt __ Do you Yahoo!? Yahoo! Finance Tax Center - File online. File on time. http://taxes.yahoo.com/filing.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New soundfiles from Allison posted
Fran Boon [EMAIL PROTECTED] wrote: I've finally uploaded the newest (LARGE) list of sound clips in .gsm format to the bugtracker. Great thanks a lot :) Assume they'll make it into CVS tarball sometime soonish. Out of interest, how much did it cost to get all those clips recorded? -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Supported USB adapters ?
Hello all, Reading the varios sources of documentation I can only find reference to one FXS USB adapter being supported by asterisk: the Digium S100U. Is there other alternatives? I´ve found that http://www.tjnet.com/ makes a voip chipset that has been used on a lot of internet phones and usb adapters. Can we use these also? I´m asking this because I´m trying to find a cheap alternative to FXS interfaces. I need one or two for a hone PBX and ATA´s are very expensive hard to find here in Brazil. Thanks, Gelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Solution== CALLERIDNAME and GotoIf -- Quoting Question
Solved my problem, but... you may not like the way I did it.
I dived into the yacc code for the humble ast_expr, and found a few
nits, and cleared them up, and submitted a bug report with a patch:
http://bugs.digium.com/bug_view_page.php?bug_id=0001292
Now, here's my complaints about the $[ ] expression parsing:
1. It uses a space, and only a space to end a token. All tokens must be
followed by exactly one space. No spaces may preceed the whole
expression (the $[ can't have a space after it). No spaces between the
last token and the closing ']'. Two spaces in a row will result in a
syntax error, most likely.
2. No variables that might contain a space are advisable. Unless of
course, it works out nicely syntax wise, which would be fairly rare, I'd
think, but who knows.
3. Double quotes are meaningless.
4. Error messages are wanting. All you get is the fact that a parse
error occurred. -- Which is good, but doesn't help you much if you
don't know what the parse error could be.
My solution:
1. Ignore leading spaces before a token.
2. Match a double quote to the next double quote, and call what's inside
a single token, spaces or no spaces. (the double quotes are not included
in the token).
3. Null tokens are ignored (trailing spaces).
4. A parse error prints out what it was parsing, and where it was in the
string, when the error occurred.
Now, this kind of thing is possible/legal and works as imagined:
exten = s,6,GotoIf($[ ${CALLERIDNUM} : 999888
${CALLERIDNAME} : Privacy Manager ]?callerid-liar|s|1:s|7)
(Pardon the obligatory inserted newline by my mail package!).
Since the evaluation pass doesn't seem to care about 's either,
the ${CALLERIDNAME} evaluates to MURPHY S , as hoped.
I've been running my new version of the expression parser for a while
now, and all seems well. I've tested both the true and false
branches. The changes I made should be upwards compatible. (Old stuff
should parse as usual.) I can't think of any cases where double quotes
might occur that things wouldn't work now... well, maybe just one, for a
null token ( parses to a null token, and will be ignored, which might
raise a syntax error... The wiki describes this and advises the use of
something like xxx${VAR} = xxx for cases like this... but you guys
know more than I do... let me know.
murf
signature.asc
Description: This is a digitally signed message part
[Asterisk-Users] T1 outgoing calls problem.
Title: Message
OK...I've got an *
box with a T100P in it. For the most part incoming calls are going through
just fine. Outgoing calls, however, I'm having some more trouble
with. Whenever I make an outgoing call, the call begins, however after the
dialing process all I hear is dead air. Here's the output from my *
console:
-- Executing
Dial("SIP/mark-2d08", "Zap/g3/2550559") in new stack --
Called g3/2550559 -- Hungup 'Zap/6-1' == Spawn
extension (uti-mainst, 2550559, 1) exited non-zero on
'SIP/mark-2d08'
I've checked with
the switch guy...and whatever channel I'm trying to dial out on is coming up as
"blocked" on his switch. We've compared as many settings as we can think
of and they all seem to be set the same. I'll post the entries from my
zaptel.conf and my zapata.conf in here...if you have any ideas please send them
my way...
zaptel.conf
span=1,1,0,d4,amiem=1-24fxsks=25loadzone=usdefaultzone=us
zapata.conf
context=outboundsignalling=em_wswitchtype=5essgroup=5callgroup=5pickupgroup=4channel
= 17-24
busydetect=yescallerid=asreceivedcallprogress=yescallreturn=yescallwaiting=yescallwaitingcallerid=yescancallforward=yesechocancel=yesechocancelwhenbridged=yesimmediate=nolanguage=usmusiconhold=defaultthreewaycalling=yestransfer=yesusecallerid=yesrelaxdtmf=no
Thanks
Mark
[Asterisk-Users] Re: 0.7.2 with cisco router 7960
yes, the 7960 is sending the right digits, because in message log from asterisk I can see each dtmf. A brief message log is below: Mar 25 19:28:33 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 (SIP/2010-9164) Mar 25 19:28:33 DEBUG[1200826048]: Bridge stops bridging channels SIP/2010-9164 and SIP/cisco2600-2b14 Mar 25 19:28:33 DEBUG[1209214528]: Difference is 976, ms is 142 Mar 25 19:28:33 DEBUG[1209214528]: Auto-deactivating generator Mar 25 19:28:33 DEBUG[1200826048]: Difference is 3192, ms is 419 Mar 25 19:28:33 DEBUG[1200826048]: Difference is 4280, ms is 555 Mar 25 19:28:38 DEBUG[1200826048]: Sending dtmf: 57 (9) Mar 25 19:28:38 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 (SIP/2010-9164) Mar 25 19:28:38 DEBUG[1200826048]: Bridge stops bridging channels SIP/2010-9164 and SIP/cisco2600-2b14 Mar 25 19:28:38 DEBUG[1209214528]: Auto-deactivating generator Mar 25 19:28:39 DEBUG[1200826048]: Difference is 3336, ms is 437 Mar 25 19:28:39 DEBUG[1200826048]: Difference is 4296, ms is 557 Mar 25 19:28:39 DEBUG[1200826048]: Sending dtmf: 50 (2) Mar 25 19:28:39 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 (SIP/2010-9164) Mar 25 19:28:39 DEBUG[1200826048]: Bridge stops bridging channels SIP/2010-9164 and SIP/cisco2600-2b14 Mar 25 19:28:39 DEBUG[1209214528]: Auto-deactivating generator Mar 25 19:28:40 DEBUG[1200826048]: Difference is 3336, ms is 437 Mar 25 19:28:40 DEBUG[1200826048]: Difference is 4296, ms is 557 Mar 25 19:28:40 DEBUG[1200826048]: Sending dtmf: 50 (2) Mar 25 19:28:40 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 (SIP/2010-9164) Mar 25 19:28:40 DEBUG[1200826048]: Bridge stops bridging channels SIP/2010-9164 and SIP/cisco2600-2b14 Mar 25 19:28:40 DEBUG[1209214528]: Difference is 2104, ms is 283 Mar 25 19:28:40 DEBUG[1209214528]: Auto-deactivating generator Mar 25 19:28:40 DEBUG[1200826048]: Difference is 2072, ms is 279 Mar 25 19:28:40 DEBUG[1200826048]: Difference is 4288, ms is 556 Mar 25 19:28:41 DEBUG[1200826048]: Sending dtmf: 56 (8) Mar 25 19:28:41 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 (SIP/2010-9164) Mar 25 19:28:41 DEBUG[1200826048]: Bridge stops bridging channels SIP/2010-9164 and SIP/cisco2600-2b14 Mar 25 19:28:41 DEBUG[1209214528]: Difference is 824, ms is 123 Mar 25 19:28:41 DEBUG[1209214528]: Auto-deactivating generator Mar 25 19:28:41 DEBUG[1200826048]: Difference is 3336, ms is 437 Mar 25 19:28:41 DEBUG[1200826048]: Difference is 4296, ms is 557 Mar 25 19:28:42 DEBUG[1200826048]: Sending dtmf: 51 (3) Mar 25 19:28:42 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 (SIP/2010-9164) Mar 25 19:28:42 DEBUG[1200826048]: Bridge stops bridging channels SIP/2010-9164 and SIP/cisco2600-2b14 Mar 25 19:28:42 DEBUG[1209214528]: Difference is 1144, ms is 163 Mar 25 19:28:42 DEBUG[1209214528]: Auto-deactivating generator Mar 25 19:28:42 DEBUG[1200826048]: Difference is 3032, ms is 399 Mar 25 19:28:42 DEBUG[1200826048]: Difference is 4296, ms is 557 Mar 25 19:28:43 DEBUG[1200826048]: Sending dtmf: 55 (7) Mar 25 19:28:43 DEBUG[1200826048]: Got AST_BRIDGE_DTMF_CHANNEL_0 on c0 (SIP/2010-9164) I tought than the wrong interpretation or transport is on Cisco 2600 when the call is outgoing and use DTMF (the voice is sending without trouble) Daniel Kurt Pasewaldt writes: Daniel, Can you determine if the 7960 is sending the right amount of digits. CME = Cisco Call Manager Express (PBX) Its a scaled down version of Call Manage and it can be ran on the following routers: 1751-v 1760 1760-v 2610XM 2611XM 2620XM 2650XM 2651XM-V 2691 3640 3640-A 3660 3725/45 IAD2420 --- Daniel Cubero Salas, Ing [EMAIL PROTECTED] wrote: Our cisco router have these dial peers: dial-peer voice 900 pots application session destination-pattern 5000 port 1/0/0 ! dial-peer voice 800 pots application session destination-pattern 9 port 1/1/1 ! dial-peer voice 701 pots application session destination-pattern 3003 port 1/0/1 ! dial-peer voice 10 pots application session destination-pattern 13T port 0/0:1-- Channelized E1 ! dial-peer voice 5 pots incoming called-number X00 direct-inward-dial ! dial-peer voice 35 pots application session destination-pattern 12T port 1/1/1 ! dial-peer voice 36 pots application session destination-pattern 14T port 1/1/0 ! dial-peer voice 1 voip application session destination-pattern ... session protocol sipv2 session target sip-server dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 4 pots incoming called-number X10 direct-inward-dial ! dial-peer voice 6 pots incoming called-number X11 direct-inward-dial ! dial-peer voice 7 pots incoming called-number X12 direct-inward-dial ! = When the call is from PSTN, detection of DTMF by Asterisk+Cisco 2600 works pretty well; but when the call is from Cisco 7960 phone thru ASTERISK+Cisco 2600 to PSTN (like IVR o PBX) always DTMF tones (for long number example 4 or more) aren´t recognized or it has wrong detection (I digit 9228373
Re: [Asterisk-Users] T1 outgoing calls problem.
Mark Messmore, Technical Support, University Telcom Inc. wrote: -- Executing Dial(SIP/mark-2d08, Zap/g3/2550559) in new stack -- Called g3/2550559 This looks like its using group 3 (g3) zapata.conf group=5 And this looks like the only group definition in zapata.conf Where is group 3 defined? does group 5 work? (g5) ? -- - Jessie Bryan Senior Systems Engineer | Network Services NetLojix Communications, Inc. e - [EMAIL PROTECTED] v - 805.884.6317 f - 805.884.6311 w - www.netlojix.com - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 outgoing calls problem.
Yeah...sorry about posting the wrong group...I've been doing a lot of testing with different groups and settings just trying to get something to work...here's group 5. context=conference signalling=em_w switchtype=5ess group=3 callgroup=3 pickupgroup=3 channel = 6 busydetect=yes callprogress=yes callreturn=yes callwaiting=yes callwaitingcallerid=yes cancallforward=yes echocancel=yes echocancelwhenbridged=yes immediate=no language=us musiconhold=default threewaycalling=yes transfer=yes usecallerid=yes Thanks for your help. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jessie Bryan Sent: Friday, March 26, 2004 2:43 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] T1 outgoing calls problem. Mark Messmore, Technical Support, University Telcom Inc. wrote: -- Executing Dial(SIP/mark-2d08, Zap/g3/2550559) in new stack -- Called g3/2550559 This looks like its using group 3 (g3) zapata.conf group=5 And this looks like the only group definition in zapata.conf Where is group 3 defined? does group 5 work? (g5) ? -- - Jessie Bryan Senior Systems Engineer | Network Services NetLojix Communications, Inc. e - [EMAIL PROTECTED] v - 805.884.6317 f - 805.884.6311 w - www.netlojix.com - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco ATA186 SIP transfer
Hello asterisk experts, I have a running installation with a Cisco 7960 and an ATA186. Attended and unattended transfer of an incoming PSTN call from 7960 to ATA works as expected. From ATA to 7960 users can press the flash button, dial the 7960, talk to the other ext. and should then be able to complete the transfer by hanging up according to Cisco's docs. Instead, the connection is dropped at 7960 when ATA hangs up and the external call rings at the 7960 like a new call. So basically transferring works, but always requires hanging up in the middle. Any ideas how to fix that? Thank you and regards, Jan Baumann [did-from-pstn] exten = 1234531,1,SetVar(ALERT_INFO=1) exten = 1234531,2,LookupCIDName exten = 1234531,3,Dial(SIP/31,20,t) exten = 1234531,4,Voicemail2(u31) exten = 1234531,5,Hangup exten = 1234531,104,Busy exten = 1234532,1,SetVar(ALERT_INFO=1) exten = 1234532,2,LookupCIDName exten = 1234532,3,Dial(SIP/32,20,t) exten = 1234532,4,Voicemail2(u32) exten = 1234532,5,Hangup exten = 1234532,104,Busy [from-sip-internal] exten = 31,1,Dial(SIP/31,30,tr) exten = 31,2,Voicemail2(u31) exten = 31,3,Hangup exten = 31,102,Busy exten = 32,1,Dial(SIP/32,30,tr) exten = 32,2,Voicemail2(u32) exten = 32,3,Hangup exten = 32,102,Busy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with Asterisk Error Please?
Can any one help me with an error im getting with
asterisk? I have VoicePulse Connect and Nufone and when i try to make a call out
on VoicePulse i get the follow error:
Mar 26 14:34:24 NOTICE[196624]: app_dial.c:545
dial_exec: Unable to create channel of type 'IAX2' == Everyone is busy
at this time -- Executing
Hangup("[EMAIL PROTECTED]:4569]/1", "") in new stack
But when i make the call out on NuFone it works
fine.
Any help would be
great.William
Re: [Asterisk-Users] Supported USB adapters ?
There is a USB handset based on the TigerJet 560/560A chip that is being marketed as a Rocket Phone... VID 06e6 PID 0210... that is the same unit as the Digium S100U. The wsfxs driver even recognizes it as the digium hardware. - Original Message - From: Gelson Dias Santos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 26, 2004 11:49 AM Subject: [Asterisk-Users] Supported USB adapters ? Hello all, Reading the varios sources of documentation I can only find reference to one FXS USB adapter being supported by asterisk: the Digium S100U. Is there other alternatives? I´ve found that http://www.tjnet.com/ makes a voip chipset that has been used on a lot of internet phones and usb adapters. Can we use these also? I´m asking this because I´m trying to find a cheap alternative to FXS interfaces. I need one or two for a hone PBX and ATA´s are very expensive hard to find here in Brazil. Thanks, Gelson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P to E100P for stress test
Dear Scott i have made every possible combination from timing source to cvs version is possible but nothing i speack a lot in chat with nice guy about it but nothing... the cable is not the problem im sure of it because i use Trend Aurora Sonata Tester (..but im not good telecom guru...im geek:-) ) and if use it to performe more test about cable and connection/timing.. So My problem is connect this type of object: - PBX with one PRI (1 span) - Box1 with 1 TE100P zaptel/libPRI latest cvs - Box1 with 1 E100P zaptel/libPRI latest cvs - Test System Sonata (1 Span) Now i want test * to obtain a LOAD TEST.. 1-Timing from Sonata Network/Master 2-Span1 (span=1,1,0,ccs,hdb3,crc4,yellow) Box1 signalling = pri_cpe 3-Span2 (span=2,1,0,ccs,hdb3,crc4,yellow) Box1 signalling = pri_cpe also try (span=2,2,0,ccs,hdb3,crc4,yellow) Box1 also try (span=2,0,0,ccs,hdb3,crc4,yellow) Box1 4-Span1 (span=1,1,0,ccs,hdb3,crc4,yellow) Box2 signalling = pri_net Passtrough call check destination SONATA-Span1 TE410P Box1 Queue the call Span2 TE410P Box1E100P Span1 Box2 This configure not work !!! About Sync, seems fail sync after 10-30 sec with Span2 Box1 ---Span1 E100P Box2 , the first time when i restart Box1 the syncing and i can call to Box2 well and works well but after some second drop with this message: Mar 22 20:30:09 WARNING[196621]: chan_zap.c:5972 zt_pri_error: PRI: Read on 77 failed: Unknown error 500 Mar 22 20:30:09 NOTICE[196621]: chan_zap.c:6687 pri_dchannel: PRI got event: 4 on span 2 Mar 22 20:30:09 WARNING[213006]: chan_zap.c:4909 handle_init_event: Detected alarm on channel 32: Yellow Alarm Mar 22 20:30:09 WARNING[213006]: chan_zap.c:4909 handle_init_event: Detected alarm on channel 33: Yellow Alarm Mar 22 20:30:09 WARNING[213006]: chan_zap.c:4909 handle_init_event: Detected alarm on channel 34: Yellow Alarm --- So after long test and coffee :-) i can only say that seems * dont manage the Network Conf is right? How is possible solution? Thanks in advance Dimitri On Thursday 25 March 2004 18:42, Scott Stingel wrote: Looks like you have to have one side of the direct connection supply a clock source. Try having box 2 source the clock on that span: span=1,1,0,ccs,hdb3,crc4,yellow Also, I've never used the Yellow option, so I don't know how that effects things. But anyway, I've done exactly what you want to do, stress test from one system to the other. Should be no problem.. Regards Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of reseaux Sent: Wednesday, March 24, 2004 6:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] TE410P to E100P for stress test Dear i have two box and i want made some stress test with one TE410P and a E100P with only one span 1 Server TE410P Span 1-- PBX Span 2---E100P Box The Box with TE410P is Mandrake 9.2 with P4 HT #zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow span=3,0,0,ccs,hdb3,crc4,yellow span=4,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 #zapata.conf group = 1 switchtype = euroisdn ;signalling = pri_net signalling = pri_cpe context=prepaid immediate=no callerid=asreceived channel = 1-15,17-31,63-77,79-93 ; ,125-139,141-155,187-201,203-217 group = 2 switchtype = euroisdn ;signalling = pri_cpe ;signalling = pri_net context=demo ;immediate=yes channel = 32-46,48-62,94-108,110-124 ; ,156-170,172-186,218-232,234-248 The second Box E100P Mandrake 9.2 P3 800Mhz #zaptel.conf /// Add PRI T100P span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 #zapata.conf group = 1 switchtype = euroisdn ;signalling = pri_net signalling = pri_net context=incoming immediate=yes callerid=asreceived ;echocancel=32 ;or yes ;echocancelwhenbridged=yes channel = 1-15,17-31 The two box works great with my lucent pbx but when i connect the two box the span i have the following error: -- Mar 25 02:41:15 NOTICE[262161]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 29 -- Mar 25 02:41:15 NOTICE[262161]: chan_zap.c:4881 handle_init_event: Alarm cleared on channel 29 -- And want sync the span... Only some time the span is syncronized and i made a one call and works but only for one or two call.. Someone can give me some hits... Thanks in advance Dimitri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
[Asterisk-Users] New astguiclient released 1.0.0
Hello, I'm pleased to announce that we are finally up to version 1.0.0 with the astguiclient suite. The biggest changes to the suite have been to make it easier to install and the addition of a complete from-scratch installation instruction document. you can see screenshots of the gui clients and download the suite here: http://astguiclient.sf.net/ Let me know what you think, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk install instructions from scratch
Hello, http://astguiclient.sourceforge.net/scratch_install.html That is the address of our new in-depth step-by-step instructions for installing Asterisk on a blank machine all the way through the installation of Linux, MySQL, Apache/PHP, Asterisk and the astguiclient suite. All with complete instructions. Take a look at it and let me know what you think, MATT--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Execute AGI application in astman
Hi, I want to add a new command to the astman interface which would allow one to call any AGI application/command using astman. As I'm new to the Asterisk source code, I would like to know if there's any caveats that I should be aware of before I start coding it. Such addition will help to increase the Asterisk's support from the CTI perspective. Thanks in advance, Raul M. Fragoso ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FreeBSD-oriented list
After the last FreeBSD-hostile response from someone on the list, I was wondering about something someone else said a few weeks ago: that something of the magnitude of a *real* Asterisk under FreeBSD project would probably require its own mailing list (and perhaps its own project website). Has anyone already setup such a thing? If not, I could lend resources towards such a project. -- Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/ BOFH Extraordiaire Sysadmin Ombudsman GPG key 0xFF676C9E GPG key fingerprint = 6272 5482 EDDD D0A3 FED2 262A FABB 599D FF67 6C9E disclaimer: My opinions are my own. Don't bother my employer about them. pgp0.pgp Description: PGP signature
[Asterisk-Users] DIAX Followup
Anyway, in my P.S. yesterday (the main post was on Codec problems), I described a situation where any IAX softphone was registering successfully, and then having zero sounds heard on either side of the call. Here is an iax2 debug output from a DIAX call to a local * server, dialing the extension that goes directly to the demo application. AsteriskHouse*CLI iax2 debug IAX2 Debugging Enabled Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 22150 DCall: 0 [10.251.1.2:4569] VERSION : 2 CALLING NUMBER : XXX-XXX- CALLING NAME: Hadar Pedhazur FORMAT : 2 CAPABILITY : 2 USERNAME: hadar CALLED NUMBER : DNID: CALLED CONTEXT : from-hadar Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 1655939482ms SCall: 4 DCall: 22150 [10.251.1.2:4569] AUTHMETHODS : 2 CHALLENGE : 133911739 USERNAME: hadar Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 000 Type: VOICE Subclass: 2 Timestamp: 00010ms SCall: 22150 DCall: 0 [10.251.1.2:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00010ms SCall: 4 DCall: 22150 [10.251.1.2:4569] Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 001 Type: IAX Subclass: AUTHREP Timestamp: 00020ms SCall: 22150 DCall: 4 [10.251.1.2:4569] MD5 RESULT : 91f6cc1e25fasd0bb43c22d366e4dcd4 -- Accepting AUTHENTICATED call from 10.251.1.2, requested format = 2, actual format = 2 -- Executing Goto([EMAIL PROTECTED]/4, default|s|1) in new stack -- Goto (default,s,1) -- Executing Wait([EMAIL PROTECTED]/4, 1) in new stack Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 003 Type: IAX Subclass: ACCEPT Timestamp: 1658659482ms SCall: 4 DCall: 22150 [10.251.1.2:4569] FORMAT : 2 Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 1658659482ms SCall: 22150 DCall: 4 [10.251.1.2:4569] -- Executing Answer([EMAIL PROTECTED]/4, ) in new stack Tx-Frame Retry[000] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: ANSWER Timestamp: 2227876761ms SCall: 4 DCall: 22150 [10.251.1.2:4569] -- Executing DigitTimeout([EMAIL PROTECTED]/4, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout([EMAIL PROTECTED]/4, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround([EMAIL PROTECTED]/4, demo-congrats) in new stack Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 003 Type: VOICE Subclass: 2 Timestamp: 2227876762ms SCall: 4 DCall: 22150 [10.251.1.2:4569] -- Playing 'demo-congrats' (language 'en') Tx-Frame Retry[001] -- OSeqno: 002 ISeqno: 003 Type: CONTROL Subclass: ANSWER Timestamp: 2227876761ms SCall: 4 DCall: 22150 [10.251.1.2:4569] Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 2227876761ms SCall: 22150 DCall: 4 [10.251.1.2:4569] Rx-Frame Retry[No] -- OSeqno: 003 ISeqno: 004 Type: IAX Subclass: PING Timestamp: 04356ms SCall: 22150 DCall: 4 [10.251.1.2:4569] Tx-Frame Retry[000] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: PONG Timestamp: 04356ms SCall: 4 DCall: 22150 [10.251.1.2:4569] Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 04356ms SCall: 22150 DCall: 4 [10.251.1.2:4569] Rx-Frame Retry[No] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: HANGUP Timestamp: 06930ms SCall: 22150 DCall: 4 [10.251.1.2:4569] CAUSE : Dumped Call Tx-Frame Retry[-01] -- OSeqno: 005 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 06930ms SCall: 4 DCall: 22150 [10.251.1.2:4569] == Spawn extension (default, s, 5) exited non-zero on '[EMAIL PROTECTED]/4' -- Hungup '[EMAIL PROTECTED]/4' AsteriskHouse*CLI iax2 no debug ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN - card? - Asterisk
I'm having trouble determining which ISDN4Linux devices are usable in the US. I want to integrate ISDN into my Asterisk PBX. My circuit provider is Qwest. Does anyone have a working ISDN BRI interface in the US? Does the fax work? Thanks, -- Michael Welter Introspect Consulting, Inc. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P to E100P for stress test
Hi Dimitri- I'm not sure, but it looks like you have too many clock sources. Assuming that the Sonata can serve as a clock source: Try this in zaptel.conf, (zapata.conf info is shown too) BOX 1: (with the TE410P) Span 1 (connected to the sonata): span=1,0,0,ccs,hdb3,crc4,yellow (and pri_cpe) Span 2 (connected to BOX 2): span=1,1,0,ccs,hdb3,crc4,yellow (and pri_net) BOX 2: (with the E100P) Span 1 (connected to BOX 1): span=1,0,0,ccs,hdb3,crc4,yellow (and pri_cpe) Note above that Box 1, span 2 is the only span to source a clock, and the only pri_net. When you use the crossover cable to connect Box 1 and Box 2, you should get green lights on the TE410P and E400P for those spans. If this doesn't work, try removing yellow from each line. Regards Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: scott at evtmedia.com URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of reseaux Sent: Thursday, March 25, 2004 8:52 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TE410P to E100P for stress test Dear Scott i have made every possible combination from timing source to cvs version is possible but nothing i speack a lot in chat with nice guy about it but nothing... the cable is not the problem im sure of it because i use Trend Aurora Sonata Tester (..but im not good telecom guru...im geek:-) ) and if use it to performe more test about cable and connection/timing.. So My problem is connect this type of object: - PBX with one PRI (1 span) - Box1 with 1 TE100P zaptel/libPRI latest cvs - Box1 with 1 E100P zaptel/libPRI latest cvs - Test System Sonata (1 Span) Now i want test * to obtain a LOAD TEST.. 1-Timing from Sonata Network/Master 2-Span1 (span=1,1,0,ccs,hdb3,crc4,yellow) Box1 signalling = pri_cpe 3-Span2 (span=2,1,0,ccs,hdb3,crc4,yellow) Box1 signalling = pri_cpe also try (span=2,2,0,ccs,hdb3,crc4,yellow) Box1 also try (span=2,0,0,ccs,hdb3,crc4,yellow) Box1 4-Span1 (span=1,1,0,ccs,hdb3,crc4,yellow) Box2 signalling = pri_net Passtrough call check destination SONATA-Span1 TE410P Box1 Queue the call Span2 TE410P Box1E100P Span1 Box2 This configure not work !!! About Sync, seems fail sync after 10-30 sec with Span2 Box1 ---Span1 E100P Box2 , the first time when i restart Box1 the syncing and i can call to Box2 well and works well but after some second drop with this message: Mar 22 20:30:09 WARNING[196621]: chan_zap.c:5972 zt_pri_error: PRI: Read on 77 failed: Unknown error 500 Mar 22 20:30:09 NOTICE[196621]: chan_zap.c:6687 pri_dchannel: PRI got event: 4 on span 2 Mar 22 20:30:09 WARNING[213006]: chan_zap.c:4909 handle_init_event: Detected alarm on channel 32: Yellow Alarm Mar 22 20:30:09 WARNING[213006]: chan_zap.c:4909 handle_init_event: Detected alarm on channel 33: Yellow Alarm Mar 22 20:30:09 WARNING[213006]: chan_zap.c:4909 handle_init_event: Detected alarm on channel 34: Yellow Alarm --- So after long test and coffee :-) i can only say that seems * dont manage the Network Conf is right? How is possible solution? Thanks in advance Dimitri On Thursday 25 March 2004 18:42, Scott Stingel wrote: Looks like you have to have one side of the direct connection supply a clock source. Try having box 2 source the clock on that span: span=1,1,0,ccs,hdb3,crc4,yellow Also, I've never used the Yellow option, so I don't know how that effects things. But anyway, I've done exactly what you want to do, stress test from one system to the other. Should be no problem.. Regards Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of reseaux Sent: Wednesday, March 24, 2004 6:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] TE410P to E100P for stress test Dear i have two box and i want made some stress test with one TE410P and a E100P with only one span 1 Server TE410P Span 1-- PBX Span 2---E100P Box The Box with TE410P is Mandrake 9.2 with P4 HT #zaptel.conf span=1,0,0,ccs,hdb3,crc4,yellow span=2,0,0,ccs,hdb3,crc4,yellow span=3,0,0,ccs,hdb3,crc4,yellow span=4,0,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 #zapata.conf group = 1 switchtype = euroisdn ;signalling = pri_net signalling = pri_cpe context=prepaid immediate=no callerid=asreceived channel = 1-15,17-31,63-77,79-93 ; ,125-139,141-155,187-201,203-217 group = 2 switchtype = euroisdn ;signalling = pri_cpe ;signalling = pri_net context=demo ;immediate=yes channel =
Re: [Asterisk-Users] ISDN - card? - Asterisk
Michael, I won't be much help because I am just a couple of steps ahead of you, but I will try. It looks like there are not many people in the U.S. using BRI. Because it is so unpopular here, there are not many cards available that work. The key to getting a card that might work here, is if the card supports the NI-1 protocol. Most cards that do support it are active cards like the Eicon Diva Server, and are not cheap. There are a few cheap passive cards that support NI-1, but I don't know if they work. Another problem with most of the cards that do work, is that they don't have a U interface, so you need to buy an external NT1, which will give you the required U interface. There is a company in Austalia that makes a card that does NI-1 and has a U interface, but I don't know if anyone has used it. The url for their card is: http://www.traverse.com.au/productview.do?product_id=14 I just picked up a cheap Dynalink card on ebay which I am going to try on Monday, when I can plug it into a BRI line. I am not holding my breath though. My suggestion is to get a more expensive, active BRI card, and see if it works. If you have any luck, please let me know. -Tor Michael Welter wrote: I'm having trouble determining which ISDN4Linux devices are usable in the US. I want to integrate ISDN into my Asterisk PBX. My circuit provider is Qwest. Does anyone have a working ISDN BRI interface in the US? Does the fax work? Thanks, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FreeBSD Segmentation Fault on start up
But I thought asterisk was just a console application, with no need for XWindows at all. Or is it on hold due to the GUI controlling mechanisms? Joe Chris Stenton wrote: Currently the asterisk port is blocked due to vulnerabilities in pwlib. Chris - Original Message - From: "Joe Lewis" [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, March 25, 2004 9:30 PM Subject: [Asterisk-Users] FreeBSD Segmentation Fault on start up To all; I've got two installations of asterisk. The last one (installed a few days ago) is from the FreeBSD ports, and many thanks, because it compiled BEAUTIFULLY! However, I can't run it. Everytime I start asterisk, I get a segmentation fault. "asterisk -c" reveals : [...snip...] [codec_gsm.so] = (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 1 == Registered translator 'lintogsm' from format SLINR to GSM, cost 5 [codec_mp3_d.so] = (MP3/PCM16 (signed linear) Translator (Decoder only)) Segmentation fault (core dumped) So, I check the core dump to see what I can find, and get : Reading symbols from /usr/local/lib/asterisk/modules/codec_gsm.so... (no debugging symbols found)...done. Loaded symbols for /usr/local/lib/asterisk/modules/codec_gsm.so Reading symbols from /usr/local/lib/asterisk/modules/codec_mp3_d.so... (no debugging symbols found)...done. Loaded symbols for /usr/local/lib/asterisk/modules/codec_mp3_d.so Reading symbols from /libexec/ld-elf.so.1...(no debugging symbols found)... done. Loaded symbols for /libexec/ld-elf.so.1 #0 0x2953ff53 in unpack_huff () from /usr/local/lib/asterisk/modules/codec_mp3_d.so (gdb) Would there, by chance, be a missing library or package that I need? Could someone point out a possible solution? (Maybe the port assumed I have an mp3 library installed?) Joe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Supported USB adapters ?
- Original Message - From: Derek Bruce [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, March 26, 2004 2:43 PM Subject: Re: [Asterisk-Users] Supported USB adapters ? The wsfxs driver even recognizes it as the digium hardware. wcusb ? Christian Hoffmeyer YottaDot Solutions Huntsville, AL (iax) 700.859.4508 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FreeBSD Segmentation Fault on start up
Joe Lewis [EMAIL PROTECTED] wrote: (Article converted from unnecessary HTML to nice plain text.) But I thought asterisk was just a console application, with no need for XWindows at all. Or is it on hold due to the GUI controlling mechanisms? You don't have to run Asterisk from the console. I start mine as a daemon, and then connect to it remotely by logging into the Asterisk box (via SSH) and typing asterisk -r. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie and Meetme configuration problem
It was in fact the problem. When I wanted to test the MeetMe feature, I have installed ZTDummy but did not recompile Asterisk program after!!! Thanks for your help. Franck -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: 25 March 2004 20:31 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie and Meetme configuration problem Mailling LIst wrote: Hi guys, I am a newbie and having problem to enter a conference room. Here is an extract of my config files: I had a look on the mailing list archive but did not find anything regarding this problem. Thanks in advance for your help This is really a FAQ. You need a Zaptel Timer. Check the Wiki, page Asterisk timer I believe. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN--asterisk-????-hylafax
hi all, I'm looking for a config to bind asterisk and hylafax. Asterisk just have to pass the line to hylafax in the dial plan. I don't have any digium hardware, but this is certainly possible. I got it with capi4hylafax, but this config is not acceptable in a large system, the call is not going through the dial plan. Thanks for your answers. Marc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Subject: [Asterisk-Users] Supported USB adapters ?
I recently purchased a tigerjet usb phone with the TIGER560B chip($40). I needed a portable phone that avoided my crappy laptop soundcard. Their salesman said the phone is supported on linux and asterisk support would be coming... When installing the linux drivers, the make crapped out. After I googled my noodle, I asked for help and they told me to install redhat(I use debian). I said they were crazy, and then they told me to fix the code, cuz I'm the linux export. On a brighter note, sales said they would RMA the phone. Adrian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Phone - Major New Release
After months of delay, I am releasing a second Beta of IAX Phone. This version looks just like the prototype I have had on display for several months. It includes a number of new features and shows signs of others that will be coming soon. New Features: - Speaker phone - Muting - Call Logging - Hold music for held calls - Registration improvements - Direct dial (by IP or machine name) - Better keyboard handling - Hide/Minimize (F12) - Restore/Show (Alt+F12) - Cool new look/feel Coming Soon: - In-Phone call recording - Phone Book or Outlook Integration - Text Messaging - Multiple Talk Paths - DND/FWD/Reject Download it at my site: http://www.sokol-associates.com Please email me or use the bug tracker on my site if/when you find issues. Take a look at the other stuff I'm working on, including an Asterisk GUI. Info and screen shots on the blog on the front page. Thanks, Steve Steven Sokol Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco ATA186 SIP transfer
Hi, -Original Message- I have a running installation with a Cisco 7960 and an ATA186. Attended and unattended transfer of an incoming PSTN call from 7960 to ATA works as expected. From ATA to 7960 users can press the flash button, dial the 7960, talk to the other ext. and should then be able to complete the transfer by hanging up according to Cisco's docs. Instead, the connection is dropped at 7960 when ATA hangs up and the external call rings at the 7960 like a new call. So basically transferring works, but always requires hanging up in the middle. Check your ATA firmware. 2.x versions used to do this with SIP. Can you try with a 3.0 firmware ? This was noted on bugs.digium.com and closed or resolved, but I can't seem to find it right now. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug 789 - Announce/Music on Hold
Hi. I have posted a fix for announce so that it does not stop the music on hold until after playing the announcement file. If you can, please test it out for me. Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Adtran TA750, any chance of working MWI ?
After some sleep, google gave me some additional information: Wade said before that MWI is done by FSK and voltage-type MWI is not supported: http://lists.digium.com/pipermail/asterisk-users/2003-August/018426.html For the existing MWI to work, voltage on the line needs to drop for a fraction of a second - I think this can happen in asterisk, probably as part of the do_monitor block where vmwi_generate is called. Does that make any sense? We have about 40 2500 sets... If I have to replace the phones themselves, does anyone have a model they'd suggest for replacing an Avaya 2500 set that supports FSK MWI? Steve On Fri, 26 Mar 2004, Steve Creel wrote: I have L36, and Onhook Messaging is enabled. Does anyone have a reference for MWI (other than that stuff that turns up on google)? Make sure that Onhook Messaging is enabled on the Adtran FXS ports. I'd also suggest upgrading to the latest 750 firmware (L36) as it fixes some specific MWI issues. -wade I have a bunch of existing ATT/Lucent/Avaya 2500YMGP sets with LED message waiting lights. Do I stand any chance of getting the Adtrans to light these? What I know so far: When you pick up the telephone, the LED flashes. If you plug two telephones in, picking up one flashes the LED on the other. Hanging the telephone up will flash the LED. Incoming calls flash the LED. Stutter dialtone is there and functioning. Debugs in chan_zap reveal that do_monitor sees there are messages, and tries to update MWI status. The Adtran is putting out 50 volts when on hook (either with or without a message waiting). If 'On Hook Messaging' is enabled (I think this is for ADSI ?), the voltage is closer to 40v. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with Asterisk Error Please?
William C. Ray [EMAIL PROTECTED] wrote: Can any one help me with an error im getting with asterisk? I have VoicePulse Connect and Nufone and when i try to make a call out on VoicePulse i get the follow error: Mar 26 14:34:24 NOTICE[196624]: app_dial.c:545 dial_exec: Unable to create channel of type 'IAX2' == Everyone is busy at this time -- Executing Hangup([EMAIL PROTECTED]:4569]/1, ) in new stack But when i make the call out on NuFone it works fine. I get that too, so I have configured my dialplan to only attempt to use VoicePulse as a backup service. I assume that when both of VoicePulse's outgoing lines are in use, they can't accept any further calls, and that message is the result. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 SIP Images
I just received my first Cisco 7960 today and was looking forward to playing with it this weekend, however I can't seem to get it working via skinny (can't find any information via the wiki regarding what needs to be on the tftp server for skinny). I would like to get my hands on the SIP images to play with it. I know I have to get a support contract through Cisco to get download access via their site which you can bet I'm going to do Monday morning, but I was hoping to work with it this weekend while I have the time. I found the release 4.4 SIP image, but it won't take due to a bug that was evidently fixed around v3.? (4k tftp buffer, and the new image is larger). At least I have a really expensive pretty phone sitting on my desk now! :-) Mitch Sharp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
