search for: vsp

Hello All, For some reason my asterisk server is not registering a port number with my VSPs. This is causing problems where people are not able to dial in from any of my SIP or IAX VSPs. I do have one VSP that has hard coded my IP and port so I can get incoming calls but this still leaves a problem with my other VSPs. Hose can I get asterisk to register my IP and port? I have b...

I'm sure this might not be the correct place to ask and I have done a Google but I can't seem to find anything that says there is a VSP that will work with * in the Ukraine. I have a friend that lives in Kiev and basically want a phone number there to be able to talk to him and have him call me. If anyone has any information on it and they are willing to share please advise. Thanks, Jeff -------------- next part...

Hi All, I think I have missed something as I am resisted with 4 VSPs and I can not dial in using any one of them using the corresponding VoIP numbers assigned with the VSP. I can make outbound calls to another VoIP number to the same provider. The weird thing is that I have a DID with a VSP and I have that working fine but try using the associate VoIP number a...

Hi All, I seem to be having a problem with all my VSPs. When I am registering with them I don't seem to be passing my port number. This problem causes other users the inability to call my VoIP number with the VSP. My VSP showed me what they are seeing. I have changed my useragent to be: Linksys/SPA941-4.1.15 Linksys/SPA941-4.1.15 Con...

...und [GoTalk] username=09xxxxxx fromuser=09xxxxxx fromdomain=sip.gotalk.com type=peer secret=xxxxxxxx qualify=yes host=sip.gotalk.com disallow=all allow=g729 ;GoTalk Inbound [09xxxxxx] username=09xxxxxx type=user secret=xxxxxxxx fromuser=09xxxxxx host=sip.gotalk.com context=from-vsp canredirect=no Registration string is register=09xxxxxx:xxxxxxxx at sip.gotalk.com/09xxxxxx David Klaverstyn Systems Administrator Information Services, Asia-Pacific Intergraph Corporation Level 3, 299 Coronation Drive Milton, QLD 4064 AU P 61.7.3510.8951 F 61.7.3510.8901 david.klaversty...

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP...

Greg, Chris, and Jay, Thanks! You've given me plenty of info to digest. I really appreciate the responses. Apologies if my list manners aren't up to snuff! Thanks again, Jason

...EHABI, but at r201326: $ cat 1.cc int f() {} $ ./bin/clang++ -target arm-linux-androideabi 1.cc -c $ readelf -u 1.o 0x0 <_Z1fv>: 0x1 [cantunwind] $ ./bin/clang++ -target arm-linux-androideabi 1.cc -c -funwind-tables $ readelf -u 1.o 0x0 <_Z1fv>: 0x8000b0b0 Compact model 0 0x00 vsp = vsp + 4 0xb0 finish 0xb0 finish

Bear with me this is a bit long winded. I am having some issues making automated outbound calls over Broadvoice from my Asterisk 1.4.2 server. For reference, none of the below issues happen when I make the calls to VoIP phones attached to the Asterisk server. What I am trying to do is call, using a .call file, out via the SIP trunk we have setup, and when the party picks up use AMD to

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I...

...my to break into the debugger when it hangs, and hopefully dump core ... But, although I *think* I've got it all, I'm obviously missing something, as it isn't breaking ... First ... I'm running a proliant server, and when I connect via SSH to ILO on that machine, and type 'vsp', I get a shell as I expect, I can type, etc ... when I reboot the machine, I get the opening splash screen with the 7(?) options (normal boot, single user mode, etc, etc) ... but I get nothing between that and the login prompt ... first sign of a problem, maybe? Next, the easy question .....

I have a phone (a panasonic globalrange phone) which always sends a fully qualified phone number. That is, for a local Canadian number, even if I key in 6135551212 it actually sends to asterisk 01116135551212. This means of course, along with "normal" phones I end up having twice as many extensions for outdialed numbers. Is there any way I could canonicalize this down to the more

...ntent-Length format And receive this error on my client side: warning: Value of ''preferred_serialization_format'' (pson) is invalid for report, user default (b64_zlib_yaml) I am getting any reports on my /var/lib/puppet/reports. -- Marley Bacelar Project Fedora Ambassador VCP, VSP. VTSP., ITILF, IBM 000-076, IBM 000-330, IBM 000-331 marleybacelar@gmail.com -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this group, send email to puppet-users@googlegroups.com. To unsubscribe from this group, send email...

...nvalid native target: XYZ (not in project) I have tried configuring like these 1. ./configure --target=XYZ 2. ./configure --target=XYZ --enable-targets=XYZ 3. ./configure --enable-targets=XYZ But every time it is not recognising the XYZ processor. What could be the problem? Thanks & Regards VSP -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20120920/d3b170bd/attachment.html>

Hi, I've Sipura 1001 phone adapter. In the settings it has separate Line 1 and Line 2 tabs, which apparently means it can control two separate phone lines. I've Asterisk@Home server and want to setup two different extensions for two phones, i.e. 201 and 202. After doing all this, I can see in Info tab that both lines are registered but only one phone gets the dials tone. Am I doing

Hi all, I've just been told from an ex workmate that my VSP (who I used to work for) has put an anti flooding limit of 80 SIP messages per IP per minute in place. I run the phone system for a facility that has a lot of extensions, but would rarely have more than 4 or 5 simultaneous external calls. Am I in danger of tripping over this limit? It soun...

...inittab]/source: No specified sources exist err: /Class[main]/Node[basenode]/Class[securetty]/File[securetty]/source: Could not describe /securetty: Fileserver module ''securetty'' not mounted Anyone can help me? Thanks in advance! -- Marley Bacelar Project Fedora Ambassador VCP, VSP. VTSP., ITILF, IBM 000-076, IBM 000-330, IBM 000-331 marleybacelar@gmail.com -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this group, send email to puppet-users@googlegroups.com. To unsubscribe from this group, send email...

....orca.data.parsers.OrcaDataSource", "openFileData","data/normclust.dat",.name="od") dsp.1 <- .JavaConstructor("DataSourcePipe",od.1,.name="dsp") sdp.1 <- .JavaConstructor("StdDataPipe",dsp.1,.name="sdp") vsp.1 <- .JavaConstructor("VarSelPipe",sdp.1,.name="vsp") rp.1 <- .JavaConstructor("Render2DPipe",vsp.1,.name="rp") wp.1 <- .JavaConstructor("WindowPipe",rp.1,.name="wp") .Java(wp.1,"reloadView") -------------- TaoExample.R...

...595, Kpfsp=0.410, Qar = 51.9, Qve = 51.9, Qgi = 12.3, Qlu = 51.9, Qbr = 3.2, Qh = 6.4, Qli = 16.5, Qk = 12.8, Qm = 7.6, Qsk = 5.0, Qad = 0.4, Qpa = 1.0, Qsp = 1.0, Var = 7.0, Vve = 14.1, Vgi = 12.4, Vlu = 1.3, Vbr = 1.3, Vh = 1.2, Vli = 12.4, Vk = 2.2, Vm = 140.0, Vsk = 49.0, Vad = 11.2, Vpa = 1.0, Vsp = 1.0 ) Fun_ODE <- function(t,y, pars){ with (as.list(c(y, pars)), { It <- dose/doseduration Car <- Aar/Var Cve <- Ave/Vve Clu <- Alu/Vlu Cli <- Ali/Vli Cbr <- Abr/Vbr Ch <- Ah/Vh Cpa <- Apa/Vpa Csp <- Asp/Vsp Cgi <- Agi/Vgi Ck <- Ak/Vk Cm <- Am/Vm Cad <-...

I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped after a few seconds ( or as soon as a command like Playback, or the call is picked up if forw...