similar to: SIP Problems with Cisco 5300 - Invalid CSeq Number

Hi all: I've no response for the last question with the same subject. Please excuse me for the extreme length of this mail, but I send 2 SIP traces. I have problem with * and 5300, when the incoming and outgoing call are routed thru the same SIP gateway (AS5300). Do I need to set an special things in sip.conf? First all, the * printout. Second, the 5300 trace. Thanks in advace, Gus

Subscribers, apply block of if statements with menu function Subscribers, For a menu: menu(c('a','b','c','d')) How to create a function that will apply to specific menu choice objects? For example: object1<-function (menuifchoices) { menu1<-menu(c('a','b','c','d')) if (menu1==1) ... menu1a<-menu... if (menu1a==1)

Hi, I am having problems with my page. I have a banner and navigation bar in controller_name.rhtml in app/views/layout. Can someone please help? Thanks In the navigation bar, I have the following links: Category1 Sub-Category1 Sub-Category2 Sub-Category3 Sub-Category4 Sub-Category5 Sub-Category6 In the main content, I have the following links: Sub-Category1 edit delete

I am trying to use chain.c32 to boot windows xp from a hard disk. In my pxelinux.cfg file I have the following: Prompt 1 Default menu1 Timeout 2 Label menu1 Kernel chain.c32 hd0 1 When my system boots it gives me the following usage error Usage: chain.c32 (hd|fd)# [partition] Boot: If I type chain.c32 hd0 1 at the boot prompt the system boots windows ok. Any help would be

i have error in install module of tor2. but it compile good. what happen ? ivr2:/usr/src/zaptel # make clean; make install rm -f torisatool makefw tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f core cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o

Hi, I wonder if anyone can help. I have a menu (Below) which works fine, the only problem is that I would like to be able to just press a key ie <1> and it go into the menu without having to press <1> and then <Enter> Has anyone got any idea on how to get it sorted? Cheers, Guy menu title PXE Boot Menu menu background graphics/main.jpg menu tabmsgrow 22 menu cmdlinerow 22 menu

I am trying to set up an auto attendant for the first time, and am having trouble getting to the submenu. My extensions.conf file looks like this: [incoming] exten=> s,1,Background,menu1 exten=> s,2,Wait,20 exten=> s,3,Goto,s|1 exten=> 1,1,Playback,option1 exten=> 2,1,Playback,option2 exten=> 3,1,Playback,option3 It is my understanding that asterisk treats the digits entered

hello, i try to build a IVR for our company my problem is that the dtmf tones are not recognized by the phones i tried several phones. BUT when i call the voicemail i can navigate with all phones through the menu. I use * 1.2 here is the context: [ivr] exten => s,1,Answer exten => s,2,SetMusicOnHold(default) exten => s,3,DigitTimeout,5 exten => s,4,ResponseTimeout,10 ;SAI menu -

Just put a new phone in place with the latest firmware from Cisco. This is the first SPA501G we have with this firmware. In the Asterisk CLI we are now seeing the error message below about once every second. When we unplug the phone, the messages quit. NOTICE[15539]: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy Thanks in advance for any assistance on this.

Szenario: UA(Grandstream) => PROXY(SER) => GATEWAY(*) => PSTN After sending the SIP ACK From Gateway (*) ACK sip:123456@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK86c0bd474ea746b5 From: "Me" <sip:123456@mydomain.de>;tag=0f63d269bc25545d To: <sip:100@mydomain.de>;tag=as05df60b5 Contact: <sip:100@192.168.0.1> Call-ID:

as far as I know there isn't. I use 80 bytes for G711U that may or may not fix your issue. You can also do a ethereal trace to find out what the actual error is. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico Alves Sent: Friday,

Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 Would anyone have an idea on how I would be able to take the mic in on the computer and put it as the "talking party" for a conference room. I would then be able to set up a "listen only" profile for others to get in on. Reason for doing this is for 'shut-in's' for my Church.

I have a server with an asterisk ss7 link connected to a Siemens working well for over a year. A few days ago I started having problems with signaling. I found the following logs in / var / log / messages Sep 12 11:49:25 call3 kernel: [1018427.030959] dahdi: Master changed to TE2/0/2 Sep 12 11:49:25 call3 kernel: [1018427.120740] dahdi: Master changed to TE2/0/1 Sep 12 11:49:26 call3 kernel:

I'm currently building a tree-like boot menu for my network, using pxelinux and vesamenu.c32. I jump from one page to another with "kernel vesamenu.c32" and "append newmenu". However, at some points I'd like to use the classic interface, for example with a distribution install image (in order to have the exact same interface as on the CD). I tried loading pxelinux.0

Hi Users, I have a problem with some of my mp3 files. they crash the system (Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to play them. Unfortunately the logs do not give me a clear fault or cause of crash but i can clearly see that ts because of the MP3 files. Its the way some files are encoded. Is there a way I can make it skip the files that can be played? I use the

<FONT face="Default Sans Serif, Verdana, Arial, Helvetica, sans-serif" size=2><div>I am using some of the sample recordings included with asterisk for my conferencing application.&nbsp; They seem to be rather choppy at times, with the overall quality not being quite where I'd like it.&nbsp; I think I saw in a previous mailing list on here that people suggested we

Oggz 0.9.1 Release ------------------ Oggz comprises liboggz and the command-line tools oggzinfo, oggzdump, oggzdiff, oggzmerge, oggzrip and oggz-validate. liboggz is a C library providing a simple programming interface for reading and writing Ogg files and streams. Ogg is an interleaving data container developed by Monty at Xiph.Org, originally to support the Ogg Vorbis audio format. This

Oggz 0.9.1 Release ------------------ Oggz comprises liboggz and the command-line tools oggzinfo, oggzdump, oggzdiff, oggzmerge, oggzrip and oggz-validate. liboggz is a C library providing a simple programming interface for reading and writing Ogg files and streams. Ogg is an interleaving data container developed by Monty at Xiph.Org, originally to support the Ogg Vorbis audio format. This

Hello, I wrote an application in wxRuby: ruby 1.8.6 wxruby 1.9.4 installed from gem windows xp sp2 After several minutes of running it crashes with the following error: "c:/ruby/lib/ruby/site_ruby/1.8/rubygems/custom_require.rb:27: [BUG] Segmentation fault ruby 1.8.6 (2007-03-13) [i386-mswin32] This application has requested the Runtime to terminate it in an unusual way. Please contact