similar to: Asterisk and SMP

Displaying 20 results from an estimated 2000 matches similar to: "Asterisk and SMP"

2004 Dec 14
3
Problems with app_realtime
It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql> describe
2004 Dec 24
3
Preventing Asterisk from sending 'h' across to SIP Provider
Hi, I want to prevent Asterisk from sending the h extension across to the SIP provider or to prevent it from hitting the script at all. The SIP Provider does not know what to do with the h extensions once it receives it. My SIP Provider takes all digits and forwards them off to a softswitch for processing. Everytime a call hangs up, it complains about running AGI scripts on hungup
2004 Jun 30
7
Asterisk Causing Cisco 7200 Router to Crash?
Hi, We are having an issue here. It seems that whenever we initialize Asterisk on our network, the router that the Asterisk server is connected to (Cisco 7200) crashes and loses it configuration. This has happended five times and each time we have tested it, it is always when Asterisk starts up. Has anyone else seen this problem? It is very odd because this is a very good router and we
2005 Jan 10
1
Ramifications of Multiple Sip Reloads Within Minutes?
We have the ability to create random UID's on own system through a custom CGI API. These UID's are written to individual sip configuration files based on the account name, so for instance sip_TEST.conf, sip_TEST2.conf, and sip_TEST3.conf, etc. Many of these UID's are created on the fly and at random times throughout the day. Right now, I have it setup to do a reload every night.
2005 Jan 12
2
Trouble building appradius
I am having trouble building appradius from http://appradius.minitelecom.org/ I configure, make, make install cpprad-1.0, but when I configure, then make appradius I get :- obelix:/usr/src/appradius/appradius1.0 # make make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib' make[1]: Nothing to be done for `all'. make[1]: Leaving directory
2004 Dec 06
1
SIP status lagged
Hi, When I do a sip show peers in the cli, the status is lagged. This peer its behind a satellite link with 600/900ms of delay. May I change some parameter in the Asterisk? Some times I cant make a phone call from the remote site to my central site. Thanks Este mensaje ha sido analizado por C4I Mail Server en busca de virus y otros contenidos peligrosos, y se considera que esta limpio.
2005 Jan 17
1
ASTCC single stage + no access number + auth usingsip username and password
> I would like to have all SIP phones to work on prepaid basis > and without need to dial any access number, instead I would > like to use the phone as normal dialing only the destination > number, for example 00464090510. I use the AccountCode for authentication. This is how, for example: exten => _00XX.,1,DeadAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2}) > Once the call is
2004 Jun 25
2
Problems Compiling and Loading asterisk-oh323 0.6.2
Hi, I having a problem compiling the wrapper for oh323. I am running Debian, kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6 and the openh323 version I have is 1.13.5. I execute the following commands first before attempting to compile the wrapper: pwlib_1.6.6: make both openh323 1.13.5 ./configure make opt asterisk-oh323 0.6.2 make
2004 Sep 14
2
3-way calling
I need to implement a procedure for creating a 3-way call, similar to what you get from the telephone company. You're in a call, you flash hook to get the switch's attention, you dial the 3rd party, you flash again to create the 3-way call. In the asterisk world, the flash would be replaced with the *+(some key). Is this implemented? How would I configure this? Thanks for any help,
2004 Sep 28
1
Sep 28 17:52:28 WARNING[163850]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 2bda648025dbf8c52fd293515d98d2c2@216.252.176.45 for seqno 102 (Non-critical Request)
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2004 Dec 28
1
Sending e-mail from dialplan
I would like help with a "dial plan" that will do the following: I feel pretty good about each of the elements except; how to e-mail the recorded file to an e-mail address. Allow a caller to call into the system: 1. Answer 2. play a short pre defined greeting 3. Allow caller to enter "PIN" during the Item #2 greeting a. If the caller entered THE valid pin (1 system
2005 Jan 03
9
Just saw your [Asterisk] xJack Segfault in Asterisk
Hi: Just saw your post while trying to solve a similar asterisk problem. Did not see any responses. Was your problem solved and what was the solution? Carey
2004 Dec 14
3
Realtime problem
I'm having trouble with the Realtime setup. I've followed the instructions on voip-info using odbc but I get this message during asterisk boot: Parsing '/etc/asterisk/sip.conf': Not found (No such file or directory) Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load config sip.conf, SIP disabled == Registered channel type 'SIP' (Session
2004 Nov 30
5
cisco dial-peer voip
I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN
2004 Jun 29
1
Registration of H323 Endpoints?
Hi, I am using the asterisk-oh323 wrapper and I am looking to allow registration of h323 endpoints and allow Asterisk to act as a gateway. The idea is simple: H323 endpoints would register with Asterisk. They each would have their own internal extension (like SIP). If a H323 endpoint dials an outbound extension, then the h323 call gets routed to a H323 Gatekeeper which then terminates
2004 Jun 30
1
Null Pointer Reference h225_1.cxx
Hi, I get this error when trying to dial an outbound extension from a sip phone: -- snip -- -- Executing Dial("SIP/2003-02d1", "OH323/3215435249@h323gk|20") in new stack -- H.323 call to 3215435249@h323gk with codec ALAW -- Called 3215435249@h323gk 0:33.283 H225 Caller:8143908 PWLib Assertion fail: Null pointer reference, file
2006 Sep 29
2
hcc not found, rcmd build
Working under Windows XP, I am compiling a package called 'pgirmess' with the command rcmd build --binary --auto-zip pgirmess I have this message error after having listed: functions text html latex example chm .... zipping help file hcc: not found cp: cannot stat 'c:/TEMP/Rbuild365620874/pgirmess/chm/pgirmess.chm': No such file or directory make[1]: *** [chm-pgirmess] Error 1
2008 Sep 19
1
readRegistry function (PR#12937)
Full_Name: Zivan Karaman Version: 2.7.2 OS: Windows XP Submission from: (NULL) (195.6.68.214) I'm puzzled by the readRegistry function. Shouldn't the "hive" argument be something like c("HLM", "HCR", "HCU", "HU", "HCC", "HPD") rather than c("HLM", "HCR", "HCU", "HU",
2017 Feb 07
2
Clang option to provide list of target-subarchs.
There are at least four clang frontends for offloading to accelerators: 1 Cuda clang 2 OpenMP 3 HCC and 4 OpenCL. These frontends will want to embed object code for multiple offload targets into a single application binary to provide portability across different subarchitectures (e.g. sm_35, sm_50) and across different architectures (e.g nvptx64,amdgcn). Problem: Different frontends
2009 Mar 17
1
Adding labels to heatmaps from image()
Hi, I have been trying to add labels to the rows of a heatmap produced using image() function. It is simply not working. Here is what I did. A2Rplot.hclust(hcc,k=length(num),col.up="black",col.down=num,lty.up=2,lty.down=1,lwd.up=1,lwd.down=2,show.labels=FALSE) #used the above external program to create a colored dendrogram xsort <- x[1:nrow(x), hcc$labels[hcc$order]]