RE: [Asterisk-Users] Bad echo on outgoing calls
Yes - the aggressive suppressor does tend to clip speech although I don't think it is half duplex. The MEC3 echo suppressor seemed to be heading in the right direction but last time I tried it it went funny after a while causing speech interruption. Iain --On Saturday, November 15, 2003 16:23:00 -0800 Ed Rubright [EMAIL PROTECTED] wrote: There was a comment made last week in this list that with echo cancellation set as MARK2 and aggressive suppressor enabled the line would no longer be full duplex! Has anyone actually noticed this? If so, does it actually cause a problem during a normal conversation? Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Gillham Sent: Saturday, November 15, 2003 1:59 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Bad echo on outgoing calls Andrew Joakimsen wrote: The X100P cards have horrible echo problems. I've heard talk about this being fixed, but havent seen anything done about it. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Larry D. Black Sent: Saturday, November 15, 2003 3:55 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Bad echo on outgoing calls I have just installed and configured asterisk I have been playing with software phones and an analog phone plugged into a TDM card. I have one line coming in on a X100P card. My X100P works quite well if I don't adjust the gain. Unfortunately it is a bit on the quiet side without the adjustment. I'll test it out with the echotraining and the gain settings. In the past with gain enabled, the echo would correct after 5-10 seconds of conversation. This is with MEC2, and I tested with and without the aggressive suppressor. -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wcfxo installatio n error
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[Asterisk-Users] wcfxo installation error
hi, i got he following error while trying to install digium cards in red hat linux 7.3. please help. [EMAIL PROTECTED] root]# modprobe zaptel lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol cpu_raise_softirq_Rd01f3ee8 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_output_wakeup_Rcb5deb89 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol register_chrdev_R8fd899d1 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol alloc_skb_R165836f4 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __pollwait_R6023e4d1 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol kmalloc_R93d4cfe6 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __wake_up_R127fda83 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol unregister_chrdev_Rc192d491 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_input_error_Rc01ed339 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol request_module_R27e4dc04 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __generic_copy_to_user_Rd523fdd3 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol proc_mkdir_R795453d8 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __generic_copy_from_user_R116166aa lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __tasklet_schedule_Red5c73bf lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol add_wait_queue_Rf32104e5 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_input_R0503a254 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __kfree_skb_R0bfb9e98 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol remove_proc_entry_R575fdf93 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol softnet_data_Rbf4543e4 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_channel_index_Rde2c3c88 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol tasklet_init_Ra5808bbf lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol sprintf_R1d26aa98 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol skb_over_panic_R24a60296 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol tasklet_kill_R79ad224b lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol schedule_R4292364c lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol kfree_R037a0cba lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol printk_R1b7d4074 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_register_channel_R84e17db1 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol remove_wait_queue_Rbb22052a lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_unit_number_Raf412d60 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol do_BUG_R577f4bff lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol create_proc_entry_Rb5bdb616 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_unregister_channel_R80c02082 lib/modules/2.4.18-3bigmem/misc/zaptel.o: insmod lib/modules/2.4.18-3bigmem/misc/zaptel.o failed lib/modules/2.4.18-3bigmem/misc/zaptel.o: insmod zaptel failed thanks C M = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wcfxo installation error
It appears you don't have the same modversions.h file as your kernel was compiled with. Search the archives for messages like http://lists.digium.com/pipermail/asterisk-users/2003-February/007588.html On Sun, 2003-11-16 at 06:43, C M wrote: hi, i got he following error while trying to install digium cards in red hat linux 7.3. please help. [EMAIL PROTECTED] root]# modprobe zaptel lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol cpu_raise_softirq_Rd01f3ee8 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_output_wakeup_Rcb5deb89 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol register_chrdev_R8fd899d1 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol alloc_skb_R165836f4 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __pollwait_R6023e4d1 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol kmalloc_R93d4cfe6 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __wake_up_R127fda83 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol unregister_chrdev_Rc192d491 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_input_error_Rc01ed339 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol request_module_R27e4dc04 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __generic_copy_to_user_Rd523fdd3 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol proc_mkdir_R795453d8 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __generic_copy_from_user_R116166aa lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __tasklet_schedule_Red5c73bf lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol add_wait_queue_Rf32104e5 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_input_R0503a254 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol __kfree_skb_R0bfb9e98 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol remove_proc_entry_R575fdf93 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol softnet_data_Rbf4543e4 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_channel_index_Rde2c3c88 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol tasklet_init_Ra5808bbf lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol sprintf_R1d26aa98 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol skb_over_panic_R24a60296 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol tasklet_kill_R79ad224b lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol schedule_R4292364c lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol kfree_R037a0cba lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol printk_R1b7d4074 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_register_channel_R84e17db1 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol remove_wait_queue_Rbb22052a lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_unit_number_Raf412d60 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol do_BUG_R577f4bff lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol create_proc_entry_Rb5bdb616 lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved symbol ppp_unregister_channel_R80c02082 lib/modules/2.4.18-3bigmem/misc/zaptel.o: insmod lib/modules/2.4.18-3bigmem/misc/zaptel.o failed lib/modules/2.4.18-3bigmem/misc/zaptel.o: insmod zaptel failed thanks C M = Designs __ Do you Yahoo!? Protect your identity with Yahoo! Mail AddressGuard http://antispam.yahoo.com/whatsnewfree ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * is crashing, when the call is accepted (H.323 - SIP)
I'v got the following scenario: Two clients (ohphone) are calling (one
at a time) the host with asterisk, which then connects to the SIP
client.
One of these hosts let's asterisk crash with a segmentation fault (i can
provide the core file, if needed) in the second, the SIP client accepts
the call. However .. if that client get's to the voicemail instead,
because the SIP client is offline, asterisk will not crash.
---
-- (client output)
wimpy is calling host x.x.x.x
Command ? Ringing phone for marlow [x.x.x.x] ...
Started logical channel: sending GSM-06.10{sw} 1
Call with marlow [x.x.x.x] established.
Started logical channel: receiving GSM-06.10{sw} 1
0:18.204 H323 Cleaner assert.cxx(105) PWLib
Assertion fail: Jitter buffer thread did not terminate, file jitter.cxx,
line 259, Error=4
---
-- (* output)
-- Executing Ringing(H323:28231, ) in new stack
-- Executing Wait(H323:28231, 2) in new stack
-- Executing Dial(H323:28231, SIP/266|15) in new stack
-- Called 266
-- SIP/266-8d50 is ringing
-- SIP/266-8d50 answered H323:28231
Segmentation fault
---
I do have a coredump from *, if that is needed.
Client (crashing *):
Slackware
libc6 2.3.2
PWLib 1.5.2
OpenH323 1.12.2
OhPhone 1.4.1
Client (not crashing *):
Debian Sid
libc6 2.3.2.ds1-10
PwLib 1.5.2 (libpt-1.5.2)
OpenH323 1.12.2 (libopenh323-1.12.2)
OhPhone 1.4.1-1
* Host:
Debian sarge/sid mix
libc6 2.3.2.ds1-8
PwLib 1.5.2 (libpt-1.5.2-2)
OpenH323 1.12.2 (libopenh323-1.12.2-3)
asterisk 0.5.0-2
asterisk-oh323-0.5.6
Client receiving the call:
libc6 2.3.2.ds1-8
kphone 3.11-1 (Debian)
Any suggestions how this can be prevented ?
What additional info is needed ?
Regards,
Martin List-Petersen
martin at list-petersen dot se
--
New systems generate new problems.
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Re: [Asterisk-Users] Bad echo on outgoing calls
The X100P cards have horrible echo problems. I've heard talk about this being fixed, but havent seen anything done about it. Depends on the installation; I have a half dozen of these cards with very very little echo problem. You might want to reverse tip and ring in your install and see if that helps; I have heard that reversed TR can really screw up echo cancellation. Also note that more echo cancel doesn't necessarily mean better echo cancel. I have echocancel=32 in my zapata.conf and it's far better than 'yes' and higher numbers. When going from my TDM400P to my X101P I seem to have had better luck with echocancelwhenbridged=yes -- you didn't mention whether you are going between two analogue interfaces or not. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bad echo on outgoing calls
PLEASE! Do *NOT* reply to a list message, erase the body, change the subject and start a new discussion! It completely destroys the list threading for people with mail clients which can properly thread messages. Isn't it far more work to do what you're doing instead of just clicking on the To: line and starting a new message? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Your thoughts..
I think there are two ways of doing it.. Either I can create an AGI that will run on the h extension and will lookup the last entry that matches the account code of the call that just ended in the MySQL CDR and calculate the call cost immediately.. Use the database. I'd recommend Postgres myself but to each their own. What will the issues be if the Master.csv is being updated at the exact moment my cron job tries to move it? is there any file locking or a method of delaying Asterisk's write or the cron's move operation till the file is availible? Cheat. do this: mv Master.csv Master.old do sleep 1 fuser Master.old /dev/null while [ $? -eq 0 ] when you mv a file (within the same filesystem) you don't change its inode; if * is accessing the file it notices nothing. Now you simply wait for * to close the file (man fuser) and it's all yours, since * will create Master.csv if it can't find it. I throw the sleep in there simply to be nice to the system. Seriously though, if you're gonna be throwing this into a database anyway, why not store it there in the first place?? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan Sequencing
But this doesn't work! As soon as we pass a number into the context, it matches successfully against _., and we get our sorry-no-match recording and the line hangs up. Here's how we force the ordering by using include to regulate order of matching: Thanks John, that's a great explanation! I do have another question regarding '.' matching and 'live' DTMF, such as me trying to do this: exten = _.,1,Dial(SIP/[EMAIL PROTECTED]) If I am dialing with a Bt101 or something that sends all the digits in a single packet, it works great. It fails miserably, however, if I'm dialing from a phone on an FXS port, or if I'm trying to do this on an answered call. I've tried experimenting with digit timeouts but that's not the solution... It'd be nice to have a number timeout where . would capture any number of digits within that timeout (or up to a # digit or something) ... Am I just not seeing the forest for the trees, or is this currently not possible? It's a royal pain in the ass to say something like _X, _XXX, _, _ and so on... Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[schaefer: Re: [Asterisk-Users] ISDN debugging and SIP dial-in issue]
On Sat, Nov 15, 2003 at 07:59:02PM +0100, Peer Oliver schmidt wrote: What is your reason to use i4l instead of the chan_capi driver (http://www.junghanns.net/asterisk/)? Did you try both, and found i4l perform better? In short: bad reason (the ability to see the AT commands). I will try CAPI ASAP. The ttyI interface would be the UNIX way to access a device, using simple AT commands and encapsulation and would make programs like vgetty happy. Unfortunately, the ttyI layer in the kernel is full of problems, especially when many communications are established or shut down (locking issues). quality. The quality is quite good. The problems are the delays. Even by cheating and diminushing the ttyI - CAPI buffers at 172 bytes there are still echo when my non asterix program does ttyI - ttyI call transfer/copying. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bad echo on outgoing calls
Sorry I was not blameing the hardware I feel it is a problem with something I am doing I am very new to this I realy want this phone to work as they are the only cost effective Hardware sip phone I have found. The echo is a local echo on the phone and the user I dial gets choppy sound. Larry D. Black CEO Black Sheep Computing, inc 2312 E Matthews Jonesboro, AR 72401 870.910.6969 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Sunday, November 16, 2003 8:25 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Bad echo on outgoing calls Also on the budgetone phone calls to other sip phones sounds like a robot and is very choppy on outgoing calls. What codec? What kind of network topology? How loaded is your LAN or WAN connection? I have zero problems with over-internet BT101 to BT101 connections, as well as BT101 to Packet8 and other SIP providers. Don't be so quick to blame the hardware; these devices are everywhere and you would think that a problem as serious as you are describing would cause a problem in selling these devices. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone International Calls
On Monday 27 October 2003 06:15 am, [EMAIL PROTECTED] wrote: TOP POSTING MADNESS continues... you need to be part of the WORLD context, and not just NANPA, otherwise 011+COUNTRY+AREA+NUMBER works as my numerous jerjer bills will testify -wasim Wasim. Can you please elaborate on this with a working example, obviously with user:password changed.' Regards...Martin -- It doesn't matter whether you win or lose -- until you lose. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming calls randomly hangup and blank calls
Hi, I have little problem and it is so embracing when u r talking to some one and line get hang-up. When some one calls from out of state or out of country my calls gets randomly hang-up with in few seconds and it happens with most of the calls. It's happening randomly I got few calls, which worked. I have also observer that with quite few call I can't hear the person and person can hear me. If they try to leave the voicemail and this problem occurs then, voicemail will hang-up on user, because it think there is no voice, so we get half voicemail of that person or blank voicemail. I don't know weather this two thinks are related to each other problems or I have two different problems. I have removed busydetect=yes and callprogress=yes then also same thing, tried to set busycount=6 but no luck. Has any one facing this kind of problem in * My setup is DID inbound line, incoming only, to Adtran TA750 (FXS-CARD)connecting to * on T1 with t100p card. Outgoing calls are working fine, since it going through normal CO lines and FXO card. Can somebody put there input, if they had faced these problems. Any help is appreciated. Regards, Kekin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Possible Bug ??Re: [Asterisk-Users] MWI and SNOM 200
After testing and playing around it seems that AST sends what is in thesip.conf:fromuser field as the VM box. Or SNOM is reading the wrong field in the SIP packet. If I set sip.conf:fromuser=*98 for my SNOM phone then when pressing MWI on that phone will ring voicemail. From looking at chan_sip.c it doesn't look like the MWI notification routines are setting this, but I'm not sure ?? On Sun, Nov 16, 2003 at 10:51:51AM +0800, Lars Boegild Thomsen wrote: I've handled it by creating an extension called 'asterisk' since the press on MWI will try to dial '[EMAIL PROTECTED]'. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Brown (CV) Sent: 15 November 2003 12:49 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MWI and SNOM 200 Hi list, how does one get a SNOM 200 MWI to work with * ?? When I press the MWI button it doesn't connect with voice mail on my * box. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] echo probs
hi all When calling (SIP|MGCP) - * - (CAPI|ZAP) - PSTN, users complain about the receiving end gets echo, especially cellular phones. Any idea why this may happen? roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan Sequencing
If I am dialing with a Bt101 or something that sends all the digits in a single packet, it works great. It fails miserably, however, if I'm dialing from a phone on an FXS port, or if I'm trying to do this on an answered call. Zap devices should handle this fine (maybe even MGCP), but SIP should fail with that sort of a configuration since we cannot differentiate between Number valid, but more could be useful and Number incomplete, therefore once we reach a match, we have to take it. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NuFone International Calls
On Sun, 16 Nov 2003, marrandy wrote:
On Monday 27 October 2003 06:15 am, [EMAIL PROTECTED] wrote:
TOP POSTING MADNESS continues...
you need to be part of the WORLD context, and not just NANPA, otherwise
011+COUNTRY+AREA+NUMBER works as my numerous jerjer bills will testify
Can you please elaborate on this with a working example, obviously with
user:password changed.'
lag just took on a whole new definition :) --
in iax.conf put
[nufone]
type=peer
secret=yourpassword
context=WORLD ; -- this bit originally stated, maynot be necessary
host=switch-1.nufone.net
disallow=all
allow=ilbc
trunk=yes
in extensions.conf put
exten = _011.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN}|60)
-
now, jerjer says you don't need CONTEXT in your iax.conf anymore, he
handles it at his end, but this is a config from donkeys ages ago and how
we used to do things in the elden days...
- wasim
p.s. for those in the know ... the super secret code name for eeks has
been herewith changed to farfon (with two dots on the o of the fon)
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Re: [Asterisk-Users] Dial Plan Sequencing
Zap devices should handle this fine (maybe even MGCP), but SIP should fail with that sort of a configuration since we cannot differentiate between Number valid, but more could be useful and Number incomplete, therefore once we reach a match, we have to take it. I can't get anything to work... two example cases: X101P answers an incoming call and I'd like to take _up to_ 7 digits. I can never get past 1 with a _., 2 with a _X., 3 with _XX. and so on... TDM400P with a regular phone plugged in... exact same problem. How do I tell * that the number's good but more digits are useful? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
Kevin wrote: Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 Don't bother. Support Asterisk and pick up a X100P from Digium. You will have more hair and less stress. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
I do support Asterisk. I have a TDM40B and X100P from Digium, I can't take the echo on the X100P. -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO Kevin wrote: Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 Don't bother. Support Asterisk and pick up a X100P from Digium. You will have more hair and less stress. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
Kevin wrote: I do support Asterisk. I have a TDM40B and X100P from Digium, I can't take the echo on the X100P. I've got dozens of X100P based systems and have only had echo trouble on 4 systems. All of them were solved by tweaking the various settings in the Zaptel Makefile and in zapata.conf or calling the telco and bitching, loudly. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
The echo issues are line or PSTN. make sure your tip and ring are correctly wired. Polarity does matter and teh X100P does not do polarity fixing like most consumer phones today. john brown chagres technologies, inc http://www.chagres.net/products/voip/ On Sun, Nov 16, 2003 at 01:35:13PM -0500, Kevin wrote: I do support Asterisk. I have a TDM40B and X100P from Digium, I can't take the echo on the X100P. -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO Kevin wrote: Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 Don't bother. Support Asterisk and pick up a X100P from Digium. You will have more hair and less stress. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
Excuse my ignorance, but could someone explain what tip and ring is and how I ensure/test that it is wired correctly? Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Brown (CV) Sent: Sunday, November 16, 2003 10:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO The echo issues are line or PSTN. make sure your tip and ring are correctly wired. Polarity does matter and teh X100P does not do polarity fixing like most consumer phones today. john brown chagres technologies, inc http://www.chagres.net/products/voip/ On Sun, Nov 16, 2003 at 01:35:13PM -0500, Kevin wrote: I do support Asterisk. I have a TDM40B and X100P from Digium, I can't take the echo on the X100P. -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO Kevin wrote: Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 Don't bother. Support Asterisk and pick up a X100P from Digium. You will have more hair and less stress. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
Just to be sure again, I did a reversal on the tip and ring with no improvement. -Original Message- From: John Brown (CV) [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO The echo issues are line or PSTN. make sure your tip and ring are correctly wired. Polarity does matter and teh X100P does not do polarity fixing like most consumer phones today. john brown chagres technologies, inc http://www.chagres.net/products/voip/ On Sun, Nov 16, 2003 at 01:35:13PM -0500, Kevin wrote: I do support Asterisk. I have a TDM40B and X100P from Digium, I can't take the echo on the X100P. -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO Kevin wrote: Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 Don't bother. Support Asterisk and pick up a X100P from Digium. You will have more hair and less stress. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
On Sun, Nov 16, 2003 at 11:11:06AM -0800, Ed Rubright wrote: Excuse my ignorance, but could someone explain what tip and ring is and how I ensure/test that it is wired correctly? In the old days, phone plugs looked like 1/4 phono jacks There was the TIP of the jack and the RING at the base of the jack. Hence TIP and RING The easiest way is to go spend $10 at your local radio shack and buy a polarity tester. TIP/RING issues also used to cause speed problems with Pre V.90 modems. Most modern modems today have auto polarity sense / fix hardware built in.. John Brown Chagres Technologies, Inc Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Brown (CV) Sent: Sunday, November 16, 2003 10:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO The echo issues are line or PSTN. make sure your tip and ring are correctly wired. Polarity does matter and teh X100P does not do polarity fixing like most consumer phones today. john brown chagres technologies, inc http://www.chagres.net/products/voip/ On Sun, Nov 16, 2003 at 01:35:13PM -0500, Kevin wrote: I do support Asterisk. I have a TDM40B and X100P from Digium, I can't take the echo on the X100P. -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO Kevin wrote: Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 Don't bother. Support Asterisk and pick up a X100P from Digium. You will have more hair and less stress. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
Ok, then I'd suggest that you need to carefully comb thru your echo settings. Keep in mind that you will need to STOP NOW your server for hardware changes. Some will disagree with this, but thats been my experience. Like JerJer, I've got boxes with X100P cards and no echo issues once I got TIP/RING and echo settings correct. john brown chagres technologies On Sun, Nov 16, 2003 at 02:18:55PM -0500, Kevin wrote: Just to be sure again, I did a reversal on the tip and ring with no improvement. -Original Message- From: John Brown (CV) [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO The echo issues are line or PSTN. make sure your tip and ring are correctly wired. Polarity does matter and teh X100P does not do polarity fixing like most consumer phones today. john brown chagres technologies, inc http://www.chagres.net/products/voip/ On Sun, Nov 16, 2003 at 01:35:13PM -0500, Kevin wrote: I do support Asterisk. I have a TDM40B and X100P from Digium, I can't take the echo on the X100P. -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO Kevin wrote: Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 Don't bother. Support Asterisk and pick up a X100P from Digium. You will have more hair and less stress. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] two X100P cards, different context
Hi, I have two X100P cards in the same system. I can use both of them to initiate and/or receive PSTN calls. I want now to define separate context for each of them, in oder to route inbound calls to different extensions. This is what I have now in zapata.conf file: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes callwaiting=yes echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=10 callerid=asreceived rxgain=10 txgain=15 channel = 1 channel = 2 What I must do to separate the two? Let's say: ZAP/1 in context 'in-number1' ZAP/2 in context 'in-number2' Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] two X100P cards, different context
based on below, you have them in the same context insert a context=foo line after channel =1 if you want channel 2 in a different context On Sun, Nov 16, 2003 at 09:32:42PM +0200, Dan wrote: Hi, I have two X100P cards in the same system. I can use both of them to initiate and/or receive PSTN calls. I want now to define separate context for each of them, in oder to route inbound calls to different extensions. This is what I have now in zapata.conf file: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes callwaiting=yes echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=10 callerid=asreceived rxgain=10 txgain=15 channel = 1 channel = 2 What I must do to separate the two? Let's say: ZAP/1 in context 'in-number1' ZAP/2 in context 'in-number2' Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
Hi John, Thanks for the info. I'll be stopping by Radio Shack to pickup a polairy tester. I have 2 X100P and 1 TDM400 card. I will be adding a SIP phone here in the next week or so. What do you recommend for the following: - What echo cancellation settings in the zaptel makefile? - What echo settings in the zapata.conf for the X100P interfaces? - What echo settings in the zapata.conf for the TDM400 interfaces? Thanks in advance for you assistance! Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Brown (CV) Sent: Sunday, November 16, 2003 11:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO On Sun, Nov 16, 2003 at 11:11:06AM -0800, Ed Rubright wrote: Excuse my ignorance, but could someone explain what tip and ring is and how I ensure/test that it is wired correctly? In the old days, phone plugs looked like 1/4 phono jacks There was the TIP of the jack and the RING at the base of the jack. Hence TIP and RING The easiest way is to go spend $10 at your local radio shack and buy a polarity tester. TIP/RING issues also used to cause speed problems with Pre V.90 modems. Most modern modems today have auto polarity sense / fix hardware built in.. John Brown Chagres Technologies, Inc Thanks, Ed -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Brown (CV) Sent: Sunday, November 16, 2003 10:54 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO The echo issues are line or PSTN. make sure your tip and ring are correctly wired. Polarity does matter and teh X100P does not do polarity fixing like most consumer phones today. john brown chagres technologies, inc http://www.chagres.net/products/voip/ On Sun, Nov 16, 2003 at 01:35:13PM -0500, Kevin wrote: I do support Asterisk. I have a TDM40B and X100P from Digium, I can't take the echo on the X100P. -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO Kevin wrote: Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 Don't bother. Support Asterisk and pick up a X100P from Digium. You will have more hair and less stress. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] two X100P cards, different context
Hi, - Original Message - From: John Brown (CV) [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November 16, 2003 9:42 PM Subject: Re: [Asterisk-Users] two X100P cards, different context based on below, you have them in the same context insert a context=foo line after channel =1 if you want channel 2 in a different context It was so simple...:-) ..Thanks a lot. Best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
On Sun, 2003-11-16 at 19:50, Jeremy McNamara wrote: I've got dozens of X100P based systems and have only had echo trouble on 4 systems. All of them were solved by tweaking the various settings in the Zaptel Makefile and in zapata.conf or calling the telco and bitching, loudly. Earlier this evening I did a complete Checkout of Zaptel and therefore put the makefile to default, afterwards I had terrific echo. With the following settings there is no echo for me on a France Telecom line with X100P to a Grandstream. # Pick your echo canceller: MARK2, MARK3, STEVE, or STEVE2 :) # #KFLAGS+=-DECHO_CAN_STEVE #KFLAGS+=-DECHO_CAN_STEVE2 #KFLAGS+=-DECHO_CAN_MARK KFLAGS+=-DECHO_CAN_MARK2 #KFLAGS+=-DECHO_CAN_MARK3 # # Uncomment for aggressive residual echo supression under # MARK2 echo canceller # KFLAGS+=-DAGGRESSIVE_SUPPRESSOR zapata.conf is rxgain = 0.0 txgain = 0.0 Hope this helps someone. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attempting to contact John Brown
I am attempting to contact John Brown from Chagres Technologies, I know he watches this list. Please contact me ASAP John, I have been trying to get hold of you for the last few weeks regarding an order but so far havent had any luck! Regards, Aaron Martin.
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Re: [Asterisk-Users] Streaming channels from Asterisk to the Internet
I have thought about doing this as well, for what may be the same application. The easiest way to do it would be to use the Console channel and audio drivers and use a mixer -- keep in mind, I'm thinking of a radio talk show, presumably with a mixer, other audio sources, etc. It would look something like this: +--+--- line out --+---++--+ POTS --| Asterisk || mixer |---| streaming server | +--+-- line in +---++--+ || | | | | CD | | | SIP Clients, Etc.Mic | Internet Etc. Where line out of the Asterisk goes to an input of the mixer and line in is connected to a monitor port on the mixer. This would be very simple to do and wouldn't require conferences. You could map inbound calls to some telephone if you wanted to screen callers or anything like that and then forward the call to the console extension when you are ready to go on the air. This would be the ideal setup, but if you have only one computer it is a bit harder. One way to do it would be to have two audio cards and loop the Asterisk output into the Icecast input, which is hard to get the audio to go back to the person on the telephone -- maybe use a conference and a SIP phone. Otherwise, maybe Icecast can be hacked a bit or glued to a sip client i.e.: sipclient sip:[EMAIL PROTECTED] | icecast for some hypothetical sip client that just listens and sends audio data to stdout. 12345 would, again, have to be a conference and there would need to be some other way of joining the conference. As far as managing the incomming call(s) you could use astman and/or queues... -w On Fri, Nov 14, 2003 at 12:18:50PM -0600, Barton Hodges wrote: Hi folks, I'm wondering if it is currently possible to configure Asterisk to forward the conversations from 2 channels into a streaming daemon, such as Icecast, so that other people connected to the Internet can listen. The concept is similar to a radio talk-show. The show host would connect to Asterisk via an X100P or through VOIP. His or her voice would then be sent to the streaming daemon for those on the Internet to hear. The show host would also have control of the other incoming channels (via a custom web-interface), which would come in via an X100P or VOIP as well. The show host and the chosen channel(s) could have a conversation streamed out to the Internet until the channel is disconnected by the host. Any input regarding the feasability of this, and the available software (such as asterisk-perl) that can be used to accomplish this would be greatly appreciated. Barton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe problem
Make sure you have at least one blank line at the bottom of your meetme.conf.. sorry but this isn't true mine doesn't... I have checked in vi If yours has drama.. what editor are you using? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
Having spent 21 years in a telephone company as an engineer, reversing tip ring will have zero impact on any 2-wire fx pstn line. The equipment in the central office (regardless of who the manufacturer happens to be) is balanced and supplies -48 volts that is fed through the outside plant to your location. The x100p card (and equivalents) do not care whether the tip and ring are reversed. If the cards really cared, they would totally fail and not just create an echo condition. (Polarity can be an issue with some other equipment, and with some types of central office trunks, but not the stuff we're talking about here.) Echo is frequently produced by external 2-wire imbalances or imperfections on the pstn line, and not by the x100p (etc) card itself. You can create an example of imbalance by placing a resister from ground to either tip or ring (not both). Some real world examples are: - wet or damaged pstn cable - bridge-taps (something done by the telco, seldom seen any more) - cheap analog phones attached to the pstn line - some expensive analog phones on the pstn line - use of lengthy untwisted inside-wire within your home/business - poor cabling techniques (inside-wire near ballast or other AC induction) - as well as many other problems external to the asterisk cards In many cases, the imperfections may be bad enough that you might even hear AC hum, noise, crosstalk, or other degradation by listening very carefully with an ordinary analog phone. (If you can hear any imperfections, the problem is rather bad.) The echo cancel function is attempting to compensate for those imperfections and imbalances. One can either muck with the echo canceling software or go find the real source of the problem. (Incidently, that's why some people are having echo problems on this list and others don't.) Given how many of our offices/labs look in terms of cabling, etc, one rather simple step to help find the source of problems is to simply run shielded twisted pair cable between the asterisk interface and the telco demarc (ensuring the shield is actually grounded), remove _all_ other cabling and phones from the line, and verify the asterisk box is running on clean AC power with an appropriate ground. (If you find the echo is gone or better, start adding those items back one at a time to see which item(s) is causing the problem. Then post it to the list so the rest of us can learn from it.) Most of the US telco's have technicians with the equipment necessary to help find the problem if the problem really is their outside plant. However, getting to that person can be a real challenge. Rich Just to be sure again, I did a reversal on the tip and ring with no improvement. -Original Message- The echo issues are line or PSTN. make sure your tip and ring are correctly wired. Polarity does matter and teh X100P does not do polarity fixing like most consumer phones today. On Sun, Nov 16, 2003 at 01:35:13PM -0500, Kevin wrote: I do support Asterisk. I have a TDM40B and X100P from Digium, I can't take the echo on the X100P. -Original Message- Kevin wrote: Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 Don't bother. Support Asterisk and pick up a X100P from Digium. You will have more hair and less stress. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
Also keep in mind if you don't come straight from the dmarc to the x100p you might have echo also: PSTN == X100P == * SERVER | | PHONE If you do the above you will get mad echo in some cases. :P I have 3 x100p's with only about 3-5 seconds of echo at the begining of the call. bkw On Sun, 16 Nov 2003, John Brown (CV) wrote: Ok, then I'd suggest that you need to carefully comb thru your echo settings. Keep in mind that you will need to STOP NOW your server for hardware changes. Some will disagree with this, but thats been my experience. Like JerJer, I've got boxes with X100P cards and no echo issues once I got TIP/RING and echo settings correct. john brown chagres technologies On Sun, Nov 16, 2003 at 02:18:55PM -0500, Kevin wrote: Just to be sure again, I did a reversal on the tip and ring with no improvement. -Original Message- From: John Brown (CV) [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO The echo issues are line or PSTN. make sure your tip and ring are correctly wired. Polarity does matter and teh X100P does not do polarity fixing like most consumer phones today. john brown chagres technologies, inc http://www.chagres.net/products/voip/ On Sun, Nov 16, 2003 at 01:35:13PM -0500, Kevin wrote: I do support Asterisk. I have a TDM40B and X100P from Digium, I can't take the echo on the X100P. -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Sunday, November 16, 2003 1:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO Kevin wrote: Has anyone seen an FXO converter for a Cisco ATA. There is someone selling a device on Ebay that claims to convert a Cisco ATA FXS port to an FXO. FX-200 VOIP PORT CONVERTER FXS to FXO http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388 Don't bother. Support Asterisk and pick up a X100P from Digium. You will have more hair and less stress. Jeremy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
Having spent 21 years in a telephone company as an engineer, reversing tip ring will have zero impact on any 2-wire fx pstn line. The equipment Why in some cases does it infact fix the echo issues? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Message lamp integration with legacy pbx during conversion
I posted this earlier on the development list. For those of you who watch both lists, please pardon the duplication. Currently,in our*lab weuseall SIP phonesso the MWI NOTIFY works perfect. I would like to do a pilot with some legacy gear, however. Accordingly, I'd like to be able to have * dial 1000X where X is the box that has a new voicemail message and 1001X when the user of mb X deletes the new message(s). The dialing should occur within the default context. In each case, our legacy gear will turn on/off the message waiting lamp. For example, if I currently dial 1000400 on my * SIP phone, the MW lamp on legacy X 400 is flipped on by the PBX. If I dial 1001400 on my * SIP phone,the MW lamp on legacy X 400 is flipped off. Does this dialing capability already exist? I would think that in any case where * would be co-implemented with another PBX, some mechanism for basic DTMF voicemail notification would be needed. Alternatively, support in voicemail.conf for a newvoicemail event command and a nonewvoicemail event command would be very helpful. I appreciate anyassistance and would also take feedback on paying for someone to contribute this functionality to *. We are really excited about using the platform in our office, and are doing so on a limited basis, but we need thissupport to move forward. Thanks. Josh
Re: [Asterisk-Users] MeetMe problem
On Sunday 16 November 2003 15:23, Brian West wrote: Make sure you have at least one blank line at the bottom of your meetme.conf.. sorry but this isn't true mine doesn't... I have checked in vi If yours has drama.. what editor are you using? What this calls to is not that you have a blank line, but that you have a trailing newline. Vi as an editor in particular always has the trailing newline and will complain if a file does not have one at startup. Other editors are not so wise as to do this. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Enhanced VoiceMail Patch... (vm2)
http://bugs.digium.com/bug_view_page.php?bug_id=156 Anyone else try this? Feedback.. gripes.. nitpicks? Please test it out and post to the bug note. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Message lamp integration with legacy pbx during conversion
On Sun, 16 Nov 2003, Josh J. Zuerner wrote: [...] For example, if I currently dial 1000400 on my * SIP phone, the MW lamp on legacy X 400 is flipped on by the PBX. If I dial 1001400 on my * SIP phone, the MW lamp on legacy X 400 is flipped off. Does this dialing capability already exist? I would think that in any case where * would be co-implemented with another PBX, some mechanism for basic DTMF voicemail notification would be needed. Alternatively, support in voicemail.conf for a newvoicemail event command and a nonewvoicemail event command would be very helpful. Just have a look at the sample.call file that comes with asterisk. You'll only have to fill out the right telephone numbers and maybe have it connect to some dummy extension that just waits a second before hanging up (if your PBX needs more than just a ring). Then put a copy of that file in /var/spool/asterisk/outgoing and * will immediately place the call you described in that file. I appreciate any assistance and would also take feedback on paying for someone to contribute this functionality to *. We are really excited about using the platform in our office, and are doing so on a limited basis, but we need this support to move forward. If you feel the urge to spend money, just contact me off-list ;-) Cheers, Siggi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo/fault isolation test gear
This is off-topic for Asterisk, but since echo problems with the X100P cards seems to be a common issue and one which people blame on telco loops, I'd suggest getting the following gear: a 3M Dynatel subscriber loop test unit. The 745 is the one I have experience with, and if you have a fairly decent clue of what you're doing, you can test line silence, resistance, voltages, polarity, etc. Well worth a few bucks. There are other models of the Dynatel line, many of which are more complex (and more expensive) but will tell you quite a bit about a copper loop. If you read this soon enough, you can bid on one right here (no, I'm not selling it): http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=2572083615category=25418 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe problem
That might just very well be it. :P On Sun, 16 Nov 2003, Tilghman Lesher wrote: On Sunday 16 November 2003 15:23, Brian West wrote: Make sure you have at least one blank line at the bottom of your meetme.conf.. sorry but this isn't true mine doesn't... I have checked in vi If yours has drama.. what editor are you using? What this calls to is not that you have a blank line, but that you have a trailing newline. Vi as an editor in particular always has the trailing newline and will complain if a file does not have one at startup. Other editors are not so wise as to do this. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO Cards in Australia
Hi All, This topic has come up before in the Asterisk mailing list many times, so I know that a lot of people have given up in waiting for a FXO card to be approved by the Australian telecommunications authority. My question is: all legalities aside - is anyone using a FXO card in Australia successfully? Thanks in advance. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attempting to contact John Brown
I replied privately back to Aaron. Seems our Spamassissin software tagged his messages as spam. With the volume of spam email I've been getting I haven't reviewed the spam folder in a bit. I've noticed a couple of other emails got tagged as well and I'll reply to those off list. john brown chagres technologies, inc +1 505 830 1200 FWD: 50870 SIP: [EMAIL PROTECTED] On Mon, Nov 17, 2003 at 10:07:55AM +1300, Aaron Martin wrote: I am attempting to contact John Brown from Chagres Technologies, I know he watches this list. Please contact me ASAP John, I have been trying to get hold of you for the last few weeks regarding an order but so far havent had any luck! Regards, Aaron Martin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Cards in Australia
On Mon, Nov 17, 2003 at 12:13:09PM +1100, Gonzalo Servat wrote: Hi All, This topic has come up before in the Asterisk mailing list many times, so I know that a lot of people have given up in waiting for a FXO card to be approved by the Australian telecommunications authority. My question is: all legalities aside - is anyone using a FXO card in Australia successfully? I have spoken to a number of Australian users who are successfully using: X100P NetJet (echo issues) AVM Fritz!Card I hope to add myself to their number shortly, since we have recieved our Fritz!es Also [EMAIL PROTECTED] seems to be having some success with the VoiceTronix openline4. All these cards are legal except the X100P. cheers, Woody PS: You are a SLUG member, no? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Cards in Australia
The answer is yes. Peter At 12:13 17/11/03 +1100, you wrote: Hi All, This topic has come up before in the Asterisk mailing list many times, so I know that a lot of people have given up in waiting for a FXO card to be approved by the Australian telecommunications authority. My question is: all legalities aside - is anyone using a FXO card in Australia successfully? Thanks in advance. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Cards in Australia
I am sure that others have used it directly... I have used it indirectly hanging off PABX extensions and even tested them on emulators... not a problem... The x100p in their current form will never pass a-tick and even c-tick might be questionable. The CE version of the card I have never seen and seeing it uses a slightly different chipset it would probably need software adjustments.. The standard veriosn has a CE symbol on it but that doesn't mean anything as only fcc details are there... If anyone has a CE version and tried it, maybe something can be done, but then I think the CE is maybe just a furfy :-) Gary. PS: PLEASE dont ask me more, I have wasted enough time on it in the past. On Mon, 17 Nov 2003 12:13:09 +1100, Gonzalo Servat wrote: Hi All, This topic has come up before in the Asterisk mailing list many times, so I know that a lot of people have given up in waiting for a FXO card to be approved by the Australian telecommunications authority. My question is: all legalities aside - is anyone using a FXO card in Australia successfully? Thanks in advance. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP alaw/ulaw
You will need to check with Cisco to see if the ATA188 has the same issues with faxing as the ATA186. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml Dave Weis wrote: Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] strange Music on Hold between SNOM, Grandstream and Asterisk
Hi List, Here is the config ext 2601 is a GS BT-101 phone ext 2062 is a SNOM 200 latest public firmware on both asterisk is Asterisk CVS-11/14/03-22:55:45 Make a call from 2601 - 2602 life good, call works have 2602 place call on hold. The music on 2601 IS NOT my music on hold. It seems its a MOH server SNOM has. take call off of hold on 2602 and 2601 still trys to play parts of the music from SNOM's server. Make a call from 2601 - 2602 life good, call works have 2601 place call on hold, SNOM plays my music but its real choppy and doesn't play well. have two GS's call each other and MOH works, not choppy, etc. So questions are: 1. how do you get the SNOM to use Asterisk as the MOH source ? 2. how does one get the music to not be choppy when a GS places a SNOM on hold john brown ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk installation error
hi, i am getting these errors while installing asterisk. i reconfigured kernel and i have all the modules installed. kernel-source readline readline-devel openssl openssl-devel this is the error: (at the last part of the installation) gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o -lresolv editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a loader.o: In function `ast_unload_resource': /usr/src/asterisk/loader.c:132: undefined reference to `dlclose' loader.o: In function `ast_load_resource': /usr/src/asterisk/loader.c:226: undefined reference to `dlopen' /usr/src/asterisk/loader.c:228: undefined reference to `dlerror' /usr/src/asterisk/loader.c:233: undefined reference to `dlsym' /usr/src/asterisk/loader.c:235: undefined reference to `dlsym' /usr/src/asterisk/loader.c:240: undefined reference to `dlsym' /usr/src/asterisk/loader.c:242: undefined reference to `dlsym' /usr/src/asterisk/loader.c:247: undefined reference to `dlsym' loader.o:/usr/src/asterisk/loader.c:249: more undefined references to `dlsym' follow loader.o: In function `ast_load_resource': /usr/src/asterisk/loader.c:282: undefined reference to `dlclose' loader.o: In function `ast_update_module_list': /usr/src/asterisk/loader.c:434: undefined reference to `pthread_mutex_trylock' channel.o: In function `ast_queue_frame': /usr/src/asterisk/channel.c:396: undefined reference to `pthread_kill' channel.o: In function `ast_do_masquerade': /usr/src/asterisk/channel.c:2112: undefined reference to `pthread_kill' channel.o: In function `tonepair_generator': /usr/src/asterisk/channel.c:2418: undefined reference to `sin' /usr/src/asterisk/channel.c:2418: undefined reference to `sin' channel.o: In function `ast_softhangup': /usr/src/asterisk/channel.c:587: undefined reference to `pthread_kill' channel.o: In function `ast_softhangup_nolock': /usr/src/asterisk/channel.c:587: undefined reference to `pthread_kill' pbx.o: In function `ast_async_goto': /usr/src/asterisk/pbx.c:1873: undefined reference to `pthread_create' pbx.o: In function `ast_pbx_outgoing_exten': /usr/src/asterisk/pbx.c:1873: undefined reference to `pthread_create' /usr/src/asterisk/pbx.c:3857: undefined reference to `pthread_create' pbx.o: In function `ast_pbx_outgoing_app': /usr/src/asterisk/pbx.c:3920: undefined reference to `pthread_create' /usr/src/asterisk/pbx.c:3952: undefined reference to `pthread_create' pbx.o:/usr/src/asterisk/pbx.c:1873: more undefined references to `pthread_create' follow callerid.o: In function `callerid_init': /usr/src/asterisk/callerid.c:97: undefined reference to `cos' /usr/src/asterisk/callerid.c:98: undefined reference to `sin' /usr/src/asterisk/callerid.c:99: undefined reference to `cos' /usr/src/asterisk/callerid.c:100: undefined reference to `sin' /usr/src/asterisk/callerid.c:101: undefined reference to `cos' /usr/src/asterisk/callerid.c:102: undefined reference to `sin' /usr/src/asterisk/callerid.c:103: undefined reference to `cos' /usr/src/asterisk/callerid.c:104: undefined reference to `sin' /usr/src/asterisk/callerid.c:105: undefined reference to `cos' /usr/src/asterisk/callerid.c:106: undefined reference to `sin' callerid.o: In function `vmwi_generate': /usr/src/asterisk/callerid.c:441: undefined reference to `rint' /usr/src/asterisk/callerid.c:441: undefined reference to `rint' /usr/src/asterisk/callerid.c:441: undefined reference to `rint' /usr/src/asterisk/callerid.c:444: undefined reference to `rint' /usr/src/asterisk/callerid.c:446: undefined reference to `rint' callerid.o:/usr/src/asterisk/callerid.c:446: more undefined references to `rint' follow tdd.o: In function `tdd_init': /usr/src/asterisk/tdd.c:70: undefined reference to `cos' /usr/src/asterisk/tdd.c:71: undefined reference to `sin' /usr/src/asterisk/tdd.c:72: undefined reference to `cos' /usr/src/asterisk/tdd.c:73: undefined reference to `sin' manager.o: In function `accept_thread': /usr/src/asterisk/manager.c:753: undefined reference to `pthread_create' manager.o: In function `init_manager': /usr/src/asterisk/manager.c:938: undefined reference to `pthread_create' asterisk.o: In function `listener': /usr/src/asterisk/asterisk.c:254: undefined reference to `pthread_create' asterisk.o: In function `ast_makesocket': /usr/src/asterisk/asterisk.c:307: undefined reference to `pthread_create' asterisk.o: In function `console_verboser': /usr/src/asterisk/asterisk.c:559: undefined reference to `pthread_kill' asterisk.o: In function `main': /usr/src/asterisk/asterisk.c:1372: undefined reference to `pthread_sigmask' /usr/src/asterisk/asterisk.c:1435: undefined reference to `pthread_sigmask' dsp.o: In function `ast_dtmf_detect_init': /usr/src/asterisk/dsp.c:213: undefined reference to `cos'
Re: [Asterisk-Users] FXO Cards in Australia
On Mon, 2003-11-17 at 12:20, Anthony Wood wrote: I have spoken to a number of Australian users who are successfully using: X100P NetJet (echo issues) AVM Fritz!Card I hope to add myself to their number shortly, since we have recieved our Fritz!es Also [EMAIL PROTECTED] seems to be having some success with the VoiceTronix openline4. All these cards are legal except the X100P. Thanks very much Anthony. VoiceTronix cards are a little out of my budget, the NatJet AVM cards are for ISDN (and we need standard analogue). PS: You are a SLUG member, no? I'm a SLUG active mailing list user, not a financial member - yet. :) Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk installation error
On Sun, Nov 16, 2003 at 08:33:22PM -0800, C M wrote: hi, i am getting these errors while installing asterisk. i reconfigured kernel and i have all the modules installed. kernel-source readline readline-devel openssl openssl-devel this is the error: (at the last part of the installation) gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o -lresolv editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a seems like the libraries it uses didn't get compiled it, this might be a automake/etc problem. You can go through manually and add -ldl -lm -lphread (might be -ptread on yours), and stuff like -lncurses (might be -lcurses) and stuff. You might have more fun by trying make clean and recompiling. loader.o: In function `ast_unload_resource': /usr/src/asterisk/loader.c:132: undefined reference to `dlclose' loader.o: In function `ast_load_resource': /usr/src/asterisk/loader.c:226: undefined reference to `dlopen' /usr/src/asterisk/loader.c:228: undefined reference to `dlerror' /usr/src/asterisk/loader.c:233: undefined reference to `dlsym' /usr/src/asterisk/loader.c:235: undefined reference to `dlsym' /usr/src/asterisk/loader.c:240: undefined reference to `dlsym' /usr/src/asterisk/loader.c:242: undefined reference to `dlsym' /usr/src/asterisk/loader.c:247: undefined reference to `dlsym' loader.o:/usr/src/asterisk/loader.c:249: more undefined references to `dlsym' follow loader.o: In function `ast_load_resource': /usr/src/asterisk/loader.c:282: undefined reference to `dlclose' loader.o: In function `ast_update_module_list': /usr/src/asterisk/loader.c:434: undefined reference to `pthread_mutex_trylock' channel.o: In function `ast_queue_frame': /usr/src/asterisk/channel.c:396: undefined reference to `pthread_kill' channel.o: In function `ast_do_masquerade': /usr/src/asterisk/channel.c:2112: undefined reference to `pthread_kill' channel.o: In function `tonepair_generator': /usr/src/asterisk/channel.c:2418: undefined reference to `sin' /usr/src/asterisk/channel.c:2418: undefined reference to `sin' channel.o: In function `ast_softhangup': /usr/src/asterisk/channel.c:587: undefined reference to `pthread_kill' channel.o: In function `ast_softhangup_nolock': /usr/src/asterisk/channel.c:587: undefined reference to `pthread_kill' pbx.o: In function `ast_async_goto': /usr/src/asterisk/pbx.c:1873: undefined reference to `pthread_create' pbx.o: In function `ast_pbx_outgoing_exten': /usr/src/asterisk/pbx.c:1873: undefined reference to `pthread_create' /usr/src/asterisk/pbx.c:3857: undefined reference to `pthread_create' pbx.o: In function `ast_pbx_outgoing_app': /usr/src/asterisk/pbx.c:3920: undefined reference to `pthread_create' /usr/src/asterisk/pbx.c:3952: undefined reference to `pthread_create' pbx.o:/usr/src/asterisk/pbx.c:1873: more undefined references to `pthread_create' follow callerid.o: In function `callerid_init': /usr/src/asterisk/callerid.c:97: undefined reference to `cos' /usr/src/asterisk/callerid.c:98: undefined reference to `sin' /usr/src/asterisk/callerid.c:99: undefined reference to `cos' /usr/src/asterisk/callerid.c:100: undefined reference to `sin' /usr/src/asterisk/callerid.c:101: undefined reference to `cos' /usr/src/asterisk/callerid.c:102: undefined reference to `sin' /usr/src/asterisk/callerid.c:103: undefined reference to `cos' /usr/src/asterisk/callerid.c:104: undefined reference to `sin' /usr/src/asterisk/callerid.c:105: undefined reference to `cos' /usr/src/asterisk/callerid.c:106: undefined reference to `sin' callerid.o: In function `vmwi_generate': /usr/src/asterisk/callerid.c:441: undefined reference to `rint' /usr/src/asterisk/callerid.c:441: undefined reference to `rint' /usr/src/asterisk/callerid.c:441: undefined reference to `rint' /usr/src/asterisk/callerid.c:444: undefined reference to `rint' /usr/src/asterisk/callerid.c:446: undefined reference to `rint' callerid.o:/usr/src/asterisk/callerid.c:446: more undefined references to `rint' follow tdd.o: In function `tdd_init': /usr/src/asterisk/tdd.c:70: undefined reference to `cos' /usr/src/asterisk/tdd.c:71: undefined reference to `sin' /usr/src/asterisk/tdd.c:72: undefined reference to `cos' /usr/src/asterisk/tdd.c:73: undefined reference to `sin' manager.o: In function `accept_thread': /usr/src/asterisk/manager.c:753: undefined reference to `pthread_create' manager.o: In function `init_manager': /usr/src/asterisk/manager.c:938: undefined reference to `pthread_create' asterisk.o: In function `listener': /usr/src/asterisk/asterisk.c:254: undefined reference to `pthread_create' asterisk.o: In function `ast_makesocket':
[Asterisk-Users] Distinctive Ring
Hi All, I was wondering what the status of distinctive ring support in Asterisk is? I had a google search read and Mark Spencer wrote some support for it. Is distinctive ring different in every country or is it pretty standard? And for my final question, does the Wildcard FXO card support distinctive ring? Essentially what I'm trying to do is route incoming calls with ring #1 to, say, 2 SIP clients and incoming calls with ring #2 to 1 SIP client, but somehow label incoming calls so the SIP client knows whether the call was for ring #1 or ring #2. Thanks in advance. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Cards in Australia
On Mon, Nov 17, 2003 at 03:49:40PM +1100, Gonzalo Servat wrote: On Mon, 2003-11-17 at 12:20, Anthony Wood wrote: I have spoken to a number of Australian users who are successfully using: X100P NetJet (echo issues) AVM Fritz!Card I hope to add myself to their number shortly, since we have recieved our Fritz!es Also [EMAIL PROTECTED] seems to be having some success with the VoiceTronix openline4. All these cards are legal except the X100P. Thanks very much Anthony. VoiceTronix cards are a little out of my budget, the NatJet AVM cards are for ISDN (and we need standard analogue). ISDN (telstra Onramp 2) is very similar in price to standard telstra lines. The only problem is you can't have ADSL ISDN on the same line. We upgraded from 2 analogue lines to 2 digital (i.e. 4 channels) for $250. But they Telstra'd up the installation so we asked for (and got) the $250 waived. It's worth thinking about it because of the Advantages of Digital signalling when using voice: Know which number was dialed Know callerid early Know when the other end has hung up Better voice quality Using Analogue with Asterisk seems to be filled with Kludges to detect hangups, busy, etc. With ISDN, the exchange does that for you. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wireless
Has anyone got a mobile wireless phone working with * yet Is it possible to use the Cisco 7920 with skinny Regards Mick West ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Distinctive Ring
I do not know the answer for #1, but for #2, I highly doubt it. What you could do is add something to the callerid to distinguish the calls. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gonzalo Servat Sent: Sunday, November 16, 2003 11:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Distinctive Ring Hi All, I was wondering what the status of distinctive ring support in Asterisk is? I had a google search read and Mark Spencer wrote some support for it. Is distinctive ring different in every country or is it pretty standard? And for my final question, does the Wildcard FXO card support distinctive ring? Essentially what I'm trying to do is route incoming calls with ring #1 to, say, 2 SIP clients and incoming calls with ring #2 to 1 SIP client, but somehow label incoming calls so the SIP client knows whether the call was for ring #1 or ring #2. Thanks in advance. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Cards in Australia
On Mon, 2003-11-17 at 16:00, Anthony Wood wrote: ISDN (telstra Onramp 2) is very similar in price to standard telstra lines. The only problem is you can't have ADSL ISDN on the same line. We upgraded from 2 analogue lines to 2 digital (i.e. 4 channels) for $250. I was a bit turned off by the $300+ installation cost. I just rang Telstra and its infact $190 if you already have a telephone line, which I do. Awesome! How come you 4 channels if you only have 2 digital lines? I thought it was one channel per line. I was told by the Telstra rep that I need a OnRamp2 which is 2 channels, 2 lines. But they Telstra'd up the installation so we asked for (and got) the $250 waived. Typical (about Telstra'ing the installation, not the setup fee discount!) It's worth thinking about it because of the Advantages of Digital signalling when using voice: Know which number was dialed Know callerid early Know when the other end has hung up Better voice quality Using Analogue with Asterisk seems to be filled with Kludges to detect hangups, busy, etc. With ISDN, the exchange does that for you. Yeah, we're now looking at it again. Local calls are pretty cheap too as long as you don't talk for too long. You mentioned echo problems with the NetJet cards. Is this still the case or was it last time you tried that it that had echo problems? I did a Google search and didn't find much on the echo problems with them. Thanks again for the good info. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FXO Cards in Australia
No No sip image for it yet Also is there any way I can change messages and extensions depending on local time ?? Also is there a way to transfer the call over PSTN if the local extension is not answered. Eg to a normal gsm mobile ?? Regards Mick West NetExpress Phone 61 08 82420173 Fax 61 08 82425099 [EMAIL PROTECTED] Disclaimer: Confidentiality: This message contains privileged and/or confidential information intended only for the use of the addressee named above. If you are not the intended recipient of this message you are hereby notified that you must not disseminate, re-transmit, copy or take any action in reliance on it. If you have received this message in error please delete the document and notify NetExpress immediately. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the views of NetExpress. The use of this Email or it's contents in any public place, eg forum, website is strictly prohibited. Viruses: Any loss/damage incurred by using this material is not the sender's responsibility. Data Actions' entire liability will be limited to resupplying the material. No warranty is made that this material is free from computer virus or any other defect. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gonzalo Servat Sent: Monday, 17 November 2003 4:03 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] FXO Cards in Australia On Mon, 2003-11-17 at 16:00, Anthony Wood wrote: ISDN (telstra Onramp 2) is very similar in price to standard telstra lines. The only problem is you can't have ADSL ISDN on the same line. We upgraded from 2 analogue lines to 2 digital (i.e. 4 channels) for $250. I was a bit turned off by the $300+ installation cost. I just rang Telstra and its infact $190 if you already have a telephone line, which I do. Awesome! How come you 4 channels if you only have 2 digital lines? I thought it was one channel per line. I was told by the Telstra rep that I need a OnRamp2 which is 2 channels, 2 lines. But they Telstra'd up the installation so we asked for (and got) the $250 waived. Typical (about Telstra'ing the installation, not the setup fee discount!) It's worth thinking about it because of the Advantages of Digital signalling when using voice: Know which number was dialed Know callerid early Know when the other end has hung up Better voice quality Using Analogue with Asterisk seems to be filled with Kludges to detect hangups, busy, etc. With ISDN, the exchange does that for you. Yeah, we're now looking at it again. Local calls are pretty cheap too as long as you don't talk for too long. You mentioned echo problems with the NetJet cards. Is this still the case or was it last time you tried that it that had echo problems? I did a Google search and didn't find much on the echo problems with them. Thanks again for the good info. Regards, Gonzalo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk installation error
On Sunday 16 November 2003 22:33, C M wrote: i am getting these errors while installing asterisk. i reconfigured kernel and i have all the modules installed. kernel-source readline readline-devel openssl openssl-devel this is the error: (at the last part of the installation) gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o -lresolv editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a loader.o: In function `ast_unload_resource': /usr/src/asterisk/loader.c:132: undefined reference to `dlclose' snip What architecture is this on? Please reply with the output of 'uname -a'. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO Cards in Australia
On Mon, Nov 17, 2003 at 04:32:50PM +1100, Gonzalo Servat wrote: On Mon, 2003-11-17 at 16:00, Anthony Wood wrote: ISDN (telstra Onramp 2) is very similar in price to standard telstra lines. The only problem is you can't have ADSL ISDN on the same line. We upgraded from 2 analogue lines to 2 digital (i.e. 4 channels) for $250. I was a bit turned off by the $300+ installation cost. I just rang Telstra and its infact $190 if you already have a telephone line, which I do. Awesome! How come you 4 channels if you only have 2 digital lines? I thought it was one channel per line. I was told by the Telstra rep that I need a OnRamp2 which is 2 channels, 2 lines. Yeah OnRamp2 replaces 1 analogue line, so we converted 2 analogue lines to 2 * OnRamp2 i.e. 4 lines. But they Telstra'd up the installation so we asked for (and got) the $250 waived. Typical (about Telstra'ing the installation, not the setup fee discount!) It's worth thinking about it because of the Advantages of Digital signalling when using voice: Know which number was dialed Know callerid early Know when the other end has hung up Better voice quality Using Analogue with Asterisk seems to be filled with Kludges to detect hangups, busy, etc. With ISDN, the exchange does that for you. Yeah, we're now looking at it again. Local calls are pretty cheap too as long as you don't talk for too long. You mentioned echo problems with the NetJet cards. Is this still the case or was it last time you tried that it that had echo problems? I did a Google search and didn't find much on the echo problems with them. There is still the problem, so bad that 4 person business I know stumped up the cash for an ISDN10 PRI install (AU$2000) and a TE410P (AU$3000) to replace a netjet ($250). I have only heard good things about the AVM Fritz!Cards with chan_capi. They are more expensive than the NetJets, but cheaper per line than the Openline4. Thanks again for the good info. I prefer Vanilla Coke to beer. :-) cheers -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Streaming channels from Asterisk to the Internet
Otherwise, maybe Icecast can be hacked a bit or glued to a sip client i.e.: sipclient sip:[EMAIL PROTECTED] | icecast for some hypothetical sip client that just listens and sends audio data to stdout. Fortunately such a SIP client actually exists: playSIP; see http://www.live.com/playSIP/ You can run (e.g.) playSIP -a sip:[EMAIL PROTECTED] | whatever (the -a option means: output the audio stream data to stdout) Ross Finlayson LIVE.COM http://www.live.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer
Does anyone know how to make Calls auto transfer to a mobile if no one answers ?? Regards Mick West ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call transfer
On Mon, 17 Nov 2003 [EMAIL PROTECTED] wrote: Does anyone know how to make Calls auto transfer to a mobile if no one answers ?? suppose your mobile number is +923008508070 exten = 15,1,Dial(IAX/farfon|30) ; try for 30 seconds on IAX exten = 15,2,Dial(Zap/1/03008508070|45) ; then try for 45 on my cell - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
