similar to: Problems with SIP

Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling forward unconditional then Asterisk accepts the call but without cname. Below is a trace. Any help is appreciated. Thanks John Bittner Simlab.net voippbx01*CLI> <-- SIP read from 216.86.35.24:63549: INVITE sip:9734333001 at 69.60.198.130:5060 SIP/2.0

I've done the unspeakable. I took a working setup of 1.0.5 and upgraded with the latest cvs. with 1.0.5 distinctive ring worked great. with the latest cvs, it doesn't seem to work with my sipura 2000 (the only thing i have to test it with). I can see in console that its sending the info, but its not working? i realize the likely hood is I messed something up somehow during checkout or

Subject says it all. How can I get the 1.3 version and 0.9.28 to compile on CentOS 5.8 ??? When I compile the two as modules I get errors. My Makefile is: obj-m += 8139cp.o 8139too.o all: make -C /lib/modules/$(shell uname -r)/build M=$(PWD) modules clean: make -C /lib/modules/$(shell uname -r)/build M=$(PWD) clean The errors I get are: Entering directory

Hi all, I need to compile alsa-project 1.0.16 on the latest centos 5.1 kernel. I am getting this error. What to do... ? CC [M] /home/silentm/MessageNet/alsa-project/alsa-driver-1.0.16/acore/sound_oss.o CC [M] /home/silentm/MessageNet/alsa-project/alsa-driver-1.0.16/acore/info_oss.o In file included from

CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/vpmadt032.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/GpakApi.o CC [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/../voicebus.o LD [M] /home/silentm/MessageNet/digium/zaptel-1.4.10/kernel/wcte12xp/wcte12xp.o

When printing a table it is broken at some point (depending how long are the associated names) >>> see example below. Is there a way to control number of columns being printed for a given chunk of the table? Best regards, Ryszard > z5 AAAAAAA BBBBBBB CCCCCCC DDDDDDD EEEEEEE FFFFFFF GGGGGGG HHHHHHH IIIIIII AAAAAAA 1.00 -0.69 -0.54 -0.88 NA NA NA

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>

I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial

Hey all, Anybody been successful running DHCPD on a GRE tunnel? When I tell DHCPD to listen on cisco1 I see this in the log Jul 23 16:21:03 atlantis dhcpd: cisco1: unknown hardware address type 778 Here is the output of ifconfig cisco1 Link encap:UNSPEC HWaddr 8B-8E-28-32-00-00-00-00-00-00-00-00-00-00-00-00 inet addr:10.199.0.2 P-t-P:10.199.0.2

Hi list! Another day, another problem... I'm checking with Nagios my Asterisk at home, and since yesterday I noticed that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours, so that I have a new IP every day), the peer of an VoIP-provider I use is UNREACHABLE. Yesterday I though it was a problem on the line, but today is the same, so I think it is something other...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hi! I have a Asterisk installation to manage my phones at home (provider is Deutsche Telekom). It works, but very often the voice is "broken"... Yesterday during a call it was very difficult to understand what my partner sayd... It can NOT be a problem of other downloads/uploads, since in that moment there were no ones... I already had the problem in the past, solved it enabling the

Hello, I tried to configure a cisco 7905 IP phone using the skinny channel but I had not much luck. The relevant portion of skinny.conf is: [cisco1] device=SEP000F3487F8E3 callerid="Alex" <123-456-789> mailbox=500 callwaiting=1 transfer=1 context=default threewaycalling=1 line => 500 ; Dial(Skinny/500@cisco1) I set up the tftp server, and prepared the following

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Am 13.06.2020 um 18:06 schrieb Michael Keuter: > So the call used Alaw as Codec. Yes, so seems it to be... It should has the better quality... But the calls done using my mobile phone in VoIP with the Asterisk have better quality as the calls done using the normal VoIP-telefon... I'm really puzzled... Luca Bertoncello (lucabert at lucabert.de)

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Hi list! I'm using Asterisk 13.14.1 from Debian 9 repositories. The provider is Deutsche Telekom und Messagenet (just for receive). I can call and receive calls, but I have a little problem: there is a "delay" of about 1-1,5 seconds between the time the voice is sent and the time when the voice is received, so that it happens very often that the peer does not get my voice and try

Hello, I get the folowing error msg in rpcclient -c 'adddriver' command: ERROR: string overflow by 1 (1024 - 1023) in safe_strcpy [adddriver "Windows NT x86" "My Driver Name 001:aaa] Printer Driver My Driver Name 001 successfully installed. My command is like this: rpcclient MYSRV -s /etc/samba/smb.conf -A auth.txt -c 'adddriver "Windows NT x86" "My

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer

Dispongo de 10 listas, cada una de ellas es, a su vez, una lista de 3 data.frame. Trato de convertirlo todo en un único data.frame. Señalo que los data.frame son de diferente número de observaciones y variables. He probado todo, y ¡zas! nada. Ruego amablemente alguna ayuda. Manuel J. [[alternative HTML version deleted]]