similar to: unistim only recognize "default" context

I'm trying chan_unistim-1.0.0.5e with asterisk-1.4.22 and Nortel i2002 phone. When I dial the numbers are not showing up on a display, and the is no frequency sound when pressing the numbers. I think it is related to chan_unistim, isn't it. Did anybody encounter this problem and/or solve it? -- #Joseph GPG KeyID: ED0E1FB7

Is anybody using Nortel IP Phone? I have (second hand) Nortel i2002 phone and when it boots I get: DHCP server unreachable F/W version: 0604D9C My setting: DHCP? [0-No, 1-Yes]: 1 DHCP: 0-Full, 1-Partial: 0 Can any body suggest how to troubleshoot it? -- #Joseph GPG KeyID: ED0E1FB7

Hello, Cedric Hans has released an UNISTIM channel driver for asterisk (stable). You can download it at : http://mlkj.net/asterisk/chan_unistim-0.9.2.tar.bz2 Copy of README : This is a channel driver for Unistim protocol. You can use at least Nortel i2004 phones with it. Only few features are supported : Send/Receive CallerID, Redial, SoftKeys, SendText(), Music On Hold, Message Waiting

Does anybody know if the process of upgrading firmware on "Linksys SPA3102-NA" in Linux is the same as on Sipura 3K as described on voip-info.org http://www.voip-info.org/wiki/view/Sipura -- #Joseph GPG KeyID: ED0E1FB7

What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux software)? I'll eventually replace this crippled Linux with something better but I don't time to play around with it as most divers and modules are still too new and not fully available in all distros. -- #Joseph GPG KeyID: ED0E1FB7

Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone. http://moziax.mozdev.org/ I tried it yesterday on eee pc, connected to asterisk on local LAN and the performance is terrible! The delay is about 2sec or 3sec. and very bad echo. I think it is the implementation of their IAX2 in their add on, as I have tried external mic. and the same delay problem. As a comparison I've tried

We have a caller ID from our phone provider "Shaw Cable" (digital phone) and it was working OK until recently. I get an error: WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have <4>, digest has <pstn-4444> NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA <sip:7804789998 at 10.10.0.103>;tag=50e17675d59121c4o1 at

Hi [Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM' [Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) I get this after I restart my asterisk 1.6, it all worked yesterday. I have the

I need another Sipura 3K and the replacement I think is Linksys SPA3102. Any input on how reliable is it? -- #Joseph GPG KeyID: ED0E1FB7

Hi, Does anybody have an updated Chan Unistim that compiles on Asterisk 1.2beta? Below is the output when compiling on Red Hat 9.0 Thanks, [root@maui2 chan_unistim-0.9.2]# make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_unistim.o

Is there any asterisk guru who can explain me how how asterisk knows which context forward the call to? -- Joseph

Hello, I am trying to configure new opus codec in asterisk 14, but unable to find any examples of codecs.conf settings for this codec. All I am trying to do - setup peer with using opus in narrow band mode (8kHz sampling rate). Does anybody know how to configure chan_opus? -- Regards, Igor Goncharovsky Unistim Dev: http://unistim.igorg.ru -------------- next part -------------- An HTML

Hi When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

Anyone know of one that I could use? Thanks, Shri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060209/51669091/attachment.htm

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

Dear R users, I'd like to change the default order of colors & line types. Especially I am using ggplot2 and using color Set1. In Set1, the default color order is red, blue, green, violet,.. ect. However, I want to put red in fourth (not first). Likewise, I want to change the order of default linetype. I want to put "solid" line in fourth. How can I do thses? R code to draw the

Hi All, During last three month I have worked on improving functionality of Nortel phones working with asterisk to replace existing Nortel station by asterisk. Many improvments done, listed below. I have only i2002 phone and unable to test if new version of channel correctly works with i2204 phone. If anyone can test it and report issues, it would be great. Please visit mantis to find out patch

hi list can U help me caller id in USTM if now working -- Starting switch on '4211 at 4211-1' to 4203 -- Executing [4203 at office:1] DumpChan("USTM/4211 at 4211-0x7f7ba4228fd0", "") in new stack Dumping Info For Channel: USTM/4211 at 4211-0x7f7ba4228fd0: ================================================================================ Info: Name=

Which file in Asterisk have a setting for time zone? When asterisk record incoming call in Master.csv the time is 6hr. ahead. I'm on: Canada/Mountain zone -- Joseph

I've Audiocodes MP-114 registered per-endpoint (2x FXO / 2x FXS) but when call comes on pstn-4444 it goes to context "fax-incoming" in sip.conf: [pstn-4444] type=friend context=incoming ... [pstn-9998] type=friend context=fax-incoming ... the device register per end point just fine, so it can find "secret=xxx" correctly but why the call is not forwarded to correct