similar to: Asterisk SIP gateway --> SCCP Phone

Displaying 19 results from an estimated 19 matches similar to: "Asterisk SIP gateway --> SCCP Phone"

2008 Jun 20
1
Voice only works from one way.
Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to
2003 Oct 18
0
DID line with Adtran TA750 and T100p
Hello, I new to this, but with the help of mailing lists archives and IRC I am able to build my PBX. Thanks to all who had help me to reach till here. I am stuck at a point where I can't find the solution on mailing lists or even on IRC. I have individual 4 DID (Direct Inward Line) coming from Telco and terminating into TA 750 to FXS card. Many of them told that Phone instrument terminates
2009 Feb 04
1
location of temporary files in deliver
deliver has the following: -- -- -- /* After buffer grows larger than this, create a temporary file to /tmp where to read the mail. */ #define MAIL_MAX_MEMORY_BUFFER (1024*128) ... static struct istream *create_raw_stream(int fd, time_t *mtime_r) ... input = i_stream_create_seekable(input_list, MAIL_MAX_MEMORY_BUFFER,
2013 Jul 16
1
pxR
Buenas tardes Una vez que he leído las variables, sexo, edad, estado civil y nacionalidad, ¿cómo genero un px para realizar las distintas consultas?. De momento lo único que he podido hacer con pxR es leer archivos de extensión px pero no sé escribirlos. Muchas gracias [[alternative HTML version deleted]]
2004 Aug 13
1
Your Amazon.com Inquiry
Greetings from Amazon.com. We''re sorry. You''ve written to an address that cannot accept incoming e-mail. But that''s OK--this automated response will direct you to the right place at Amazon.com to answer your question or help you contact customer service if you need further assistance. You will find the answers to the most common questions here: Where''s My
2020 May 18
0
ether-wake
>> -----Original Message----- >> From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Rich Greenwood >> Sent: Monday, 18 May, 2020 08:34 >> To: centos at centos.org >> Subject: Re: [CentOS] ether-wake >> >> Some switch hardware can generate the packets directly, negating the need >> for a box on every VLAN. Meraki hardware can do it, but
2005 Feb 13
2
GRE tunnel problems
Hello, Here is my network: ------------------ ------------- ----------- Linux box ----------- GRE --------- Cisco ---------- ------------------ ------------- What I wan to accomplish. I want ripv2 to go across (both ways) through the GRE tunnel. No packets are being passed thought the GRE
2004 Sep 17
1
ZAPTEL Compile Problem?
Has anyone received this message while attempting to do a MAKE INSTALL on ZAPTEL depmod: *** Unresolved symbols in /lib/modules/2.4.26-gentoo-r9/misc/ztdummy.o I am not running DEVFS so that's not the issue. Zaptel drivers/modules all load, but ztdummy won't /lib/modules/2.4.26-gentoo-r9/misc/ztdummy.o: /lib/modules/2.4.26-gentoo-r9/misc/ztdummy.o: unresolved
2010 Oct 06
3
integrate Intertel Axxess with Asterisk
Has anyone successfully integrated Asterisk with an Inter-tel Axxess phone system via a SIP trunk using the IPRC card? -- Marvin Horst -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101006/5dbe435a/attachment.htm
2003 Sep 24
0
Re: Asterisk-Users digest, Vol 1 #1380 - 15 msgs
You have the session target as the IP address of the router's own ethernet interface. You probably want that to be the address of the Asterisk server instead. I also highly recommend you use full duplex ethernet, as voice packets don't really like to be restransmitted when a collision happens. -d > Message: 10 > From: "Bartosz Jozwiak" <bartek@cq-link.sr> >
2013 Mar 06
2
Change RX Signalling Bits in Dahdi drivers
Greeting, I am trying to setup PLAR signalling in asterisk. I have modified the FXSLS TX bits in dahdi-base.c on line 2580, and I can make calls. .sig_type = DAHDI_SIG_FXSLS, .bits[DAHDI_TXSIG_ONHOOK] = DAHDI_BITS_ABCD, /*changed by for PLAR*/ .bits[DAHDI_TXSIG_OFFHOOK] = (0), /*changed by for PLAR*/ .bits[DAHDI_TXSIG_START] = DAHDI_BITS_ABCD, /*changed by for PLAR*/ When I got to change
2020 May 18
3
ether-wake
Some switch hardware can generate the packets directly, negating the need for a box on every VLAN. Meraki hardware can do it, but you have to go through the dashboard so automating it isn't currently possible. Here is some documentation on forwarding WoL on catalyst 3750 switches from Cisco:
2007 May 21
3
Originate and bridge Can it be done? Best Way?
Hi, Im new, but trying real hard! I just need general direction, not details yet..i'll try to figure those...just looking to avoid brick walls...bottlenecks...inefficiencies etc upfront. Hardware: motorola vt2442 - trixbox Apps: Dot Net application that operates the Manager API and the FASTAGI interfaces. I have the 2442 set as a PLAR so as soon as the ext is off-hook, it dials into the *61
2006 Dec 11
2
asterisk PLAR
Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown). The Oreilly (Asterisk: Future of Telephony) book mentions it in passing saying that all you need to enable it is to set immediate=yes in zapata.conf. Has anyone implemented this in brokerage trading environments? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Oct 03
2
asterisk, cisco 3640's and DIDs...
I would think I could do this but for some reason I am stymied. I have a PRI from RCN connected to a cisco 3640 (in my day "cisco" is all lower case :-)). My config looks something like this on the cisco... --------------------------------------------------------- voice-card 3 dsp services dspfarm ! ip cef ! isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode
2004 May 07
4
Concept for line appearances and bridging: anyone?
OK, here's a configuration challenge: I want to have certain line appearances able to be "interrupted" by various other line apperances elsewhere in the office. This is harder to describe than it is to demonstrate, so I'll do that: Let's assume I have Cisco 7960's on all desks. 1) Call comes from inbound line X destined on extension 1234 2) Phones A, B, C all
2005 Mar 10
2
Cisco and Asterisk
Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are
2008 Jun 30
4
Rebuild of kernel 2.6.9-67.0.20.EL failure
Hello list. I'm trying to rebuild the 2.6.9.67.0.20.EL kernel, but it fails even without modifications. How did I try it? Created a (non-root) build environment (not a mock ) Installed the kernel.scr.rpm and did a rpmbuild -ba --target=`uname -m` kernel-2.6.spec 2> prep-err.log | tee prep-out.log The build failed at the end: Processing files: kernel-xenU-devel-2.6.9-67.0.20.EL Checking
2004 Sep 15
0
No Audio in Voicemail
For some reason I get the "Comedian Mail" prompts, I can move around in mail, "listen" delete save record prompts. BUT. When I try to leave a message, I hear no greeting, it claims to be recording in the debug window, but I get this error. Sep 15 19:10:08 ERROR[180236]: sccp_actions.c:624 sccp_handle_open_recieve_channel_ack: Device SEP000BBE94840A sent