search for: mschult

Any ideas? I edited the Makefile as instructed, ty. Now compiling .... sccp_channel.c 279 lines sccp_channel.c: In function `sccp_channel_send_callinfo': sccp_channel.c:48: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no member named `callerid' sccp_channel.c:49: structure has no

We're looking at implementing Asterisk in our department in the near future, we're looking at anywhere from 15-25 extensions. The machine we were looking at running this on was a Quad Xeon 450mhz (2MB L2 Cache) w/ 1GB of ram. I've heard bad things about running Asterisk on SMP machines? Would we be running into any performance issues with this machine? Tim Jackson Network Engineer

Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple

Anyone have experience with Sipura's? Anyone know if they offer a warranty? Would like opinions on these, good or flame. We bought *one* to test with and it died, can't even get a response from Sipura "support". Could anyone recommend another device to replace these? Prefer 1 or 2 port design. Ty :-)

Having problems getting realtime working, I'm trying to use odbc for all of this. I've got Fedora 3 and have been fighting with odbc for a day now. I think I got it working correctly, however I can't seem to get the realtime portion working. In asterisk 'odbc show' shows it connected, I see it on my (odbc) mysql server connected and all, it connects and just idles. So, without

Does anyone have some decent Nagios scripts out there that do more than monitor the proc itself? Rather than reinvite the wheel, figured I'd ask. I already saw the one on the wiki. Matt

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

...d problems Hi Matt, Seems you have gone one step further than me, Can you possibly let em see your sip configs for the cisco 7960 and also the configs in the phone. I am having trouble getting it to talk to asterisk Thanks Paul ----- Original Message ----- From: "Matt Schulte" <mschulte@netlogic.net> To: <asterisk-users@lists.digium.com> Sent: Thursday, December 16, 2004 3:01 PM Subject: [Asterisk-Users] Cisco 7960 (SIP) hold problems > Has anyone had problems with using hold on a 7960 SIP firmware? The > problem is when the 7960 puts a call on hold and you take...

Has anyone come up with a way to get power to a TDM400P card installed in a Dell PowerEdge 1750? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this

Hello, I've recently received a Cisco 7960G phone with the factory default SCCP firmware on it. As we're using SIP on our network, the first things i've done was to upgrade but unfortunately the phone just restarted. By looking on the TFTP logs and tcpump output, i've seen that the phone crashed and restarted just after downloading the OS79XX.TXT file, without requesting the image

Another nibbage question: modprobe tor2 -v install /sbin/modprobe --ignore-install tor2 && /sbin/ztcfg insmod /lib/modules/2.6.5-1.358smp/misc/tor2.ko ZT_SPANCONFIG failed on span 1: No such device or address (6) FATAL: Error running install command for tor2 dmesg states that the module is registered however. Am I doing something seriously wrong? I have my zaptel.conf configured as

Are there any considerations to take in account when faxing from analog to SIP using ULAW? The problem we're having is faxes are only making it halfway, getting cut off. Neither fax machine seems to report an error. Pretty diagram: FXS --> SIP --> PSTN Provider --> FAX ^ULAW

Kind of off topic but I know CVS is the "prefered" way of upgrading, however are there such things as "stable" CVS upgrades? It seems a lot of the CVS's have a lot of devel bugs in this that I would be scared to put even near production. Just IMHO. :-) Matt -----Original Message----- From: Rich Adamson [mailto:radamson@routers.com] Sent: Friday, August 27, 2004 9:15 AM

Ok, I searched the lists and found no definitive answer. I'm assuming the IAXy has some primitive form of echo cancel, is there anyway to adjust this? Or any ideas on what to do instead. Here's the setup, this will not be a typical setup for our company however, well whatever. Anyway it looks like this: IAXy --> Netgear router --> Charter --> Level3 --> Asterisk --> VOIP or

This maybe a simple question however I can't find a way to do this, I'm wanting to EITHER: Pass SIP digest authentication via dialplan (extensions.conf) OR Make Asterisk realize that the incoming peer in sip.conf doesn't have to authenticate. The reason I have this is because I'm connecting through SER and using aliases. The reason I'm using aliases is because we're

All, We are using a SIP provider that is expecting 0-15 response for fmtp. Our CVS Head asterisk server is sending 0-16, I looked up an rfc and it stated: RTP Payloads for Telephone Signal Events RFC 2833 Henning Schulzrinne, Scott Petrack. May 2000 Implementation notes: * Implementations can support events 0 through 15 (DTMF) by simply ignoring the packets, but MUST

Does anyone know if Asterisk can accept cleartext auth (SIP), as in it recv's a call destined to: 1234:blah@har.har.com The problem I'm having is simply for faxing, normal calls come in as g729 and of course we need ULAW for faxes. sip.conf snippet [sipfarm] insecure=very host=blahblah.netlogic.net type=peer context=sip-out username=+18165551212 secret=blah canreinvite=no disallow=all

I found a few mentions of the 7914 being used with Asterisk, these all covered SCCP/skinny though. Does anyone know if the 7914 can even be used with SIP? If so, any pointers? Is it a services thing? Anyone get the operator (line/extension status) to work with it. Thanks for the help, Cisco doesn't even mention ANYTHING about SIP + the 7914. Matt

Has anyone had any success with the Rhino CB-24? I can't get mine to work, I tried all the obvious settings. The cb-24 gets stuck at init ESF framing, as if it's not seeing the t1 card at all. It does get a t1 carrier (detecting voltage??) Help! Thanks.. Everything appears to look good on the server, I did: [zaptel.conf] span=1,0,0,esf,b8zs fxogs=1-24 [zapata.conf] signalling=fxo_gs

I'm assuming asterisk does not have a SIP jitter buffer in place? Any ideas on how to help with this going over a data T1 where VoIP is shared with regular traffic? Problem is when people are downloading the voice is jittery, even lossy. Matt