search for: 7001

I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001 at 192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes. I'm not seeing anything in the messages log. I'm sure I'm doing something wrong, just not sure where to loo...

...-22 md: starting md1 failed EXT3-fs: unable to read superblock No filesystem could mount root, tried: ext3 cramfs Kernel panic - not syncing: VFS: Unable to mount root fs on unknown-block(9,0) ==================================== root@xenhost:~# cat /var/log/xen/xend.log [..] [2008-08-09 17:18:49 7001] DEBUG (XendDomainInfo:84) XendDomainInfo.create([''vm'', [''name'', ''test1''], [''memory'', ''256''], [''vcpus'', ''8''], [''on_xend_start'', ''ignore''...

...aw this. Here are some scenario's.. This one is a succesfully completed transfer (but notice the WARNING's): -- Accepting AUTHENTICATED call from 217.114.97.249, requested format = 8, actual format = 8 -- Executing Dial("IAX2[virtupbx@virtupbx]/16384", "Local/7001@pbx") in new stack -- Called 7001@pbx -- Executing Dial("Local/7001@pbx-0685,2", "MGCP/aaln/1@217.114.96.220") in new stack -- MGCP mgcp_request(aaln/1@217.114.96.220) -- MGCP cw: 0, dnd: 0, so: 0, sno: 0 -- MGCP mgcp_new(MGCP/aaln/1@217.114.96.22...

...terisk in local network for testing. My sip.conf file looks like this [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic secret=123abcd context=internal [7002] type=friend host=dynamic secret=456abcd context=internal Am using linphone as sip client and create account on linphone with user name 7001 and 7002 7001 is running on 192.168.2.15:5060 7002 is running on 192.168.2.45:5060 when i...

...t on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the calling A-Leg 7001 NativeFormats: (g722) to the B-Leg 7000 NativeFormats: (alaw) -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY ------------------------------------------------------------------------ ------------------------------------------------------------------------ -------------...

Hi, With a 1.4.35 or 1.6.1.19, I'm facing this behaviour : - extension 7002 is a SIP hard phone currently configured to forward incoming calls to extension 7003, when a call is unanswered within a 10s time frame - when extension 7001 is calling extension 7002 with a Dial(SIP/7002,20) statement and no one answers, then : - after 10s, Asterisk receives "SIP 302 Moved temporarily" message and enters its dialplan to call 7003, as required, - 10s later (or 20s from the very start), call from 7001 to 7003 is cut and the nex...

I just installed the newly released Asterisk 1.4.7 and I cannot get music on hold. I am using the default settings with the wav files. Here is what I get on the cli from any sip phone: -- Executing [7001 at oficina:1] NoCDR("SIP/1120-084e6010", "") in new stack -- Executing [7001 at oficina:2] Answer("SIP/1120-084e6010", "") in new stack -- Executing [7001 at oficina:3] MusicOnHold("SIP/1120-084e6010", "default") in new stack [Jul...

...ot;, "1, Caller "DEMKOVITCH,IVAN" <13144880983> has entered the sales queue") in new stack   Caller "aa" <15555555555> has entered the sales queue     -- Executing [1 at automated_attendant_normal:2] Goto("SIP/callcentric15-00000435", "queues,7001,1") in new stack     -- Goto (queues,7001,1)     -- Executing [7001 at queues:1] Verbose("SIP/callcentric15-00000435", "2,"aa" <1555555> entering sales queue") in new stack   == "aa" <15555555555> entering sales queue     -- Executing [7001 a...

For some time now I try to get my code working. This Monday I switched to RJS (first with 1.0 + plugin) and yesterday to Edge Rails, so I guess I''ve been using the most recent version. I have a div with id="detail" which I want to update with a partial. If I''m using page.replace_html the content will be replaced with text, i.e. the javascript won''t become

...We tend to park calls, but we're often not able to pick them back up, or the other party says they get dropped, etc. There doesn't seem to be a specific pattern that I've discovered so far. I had this happen to me personally this morning -- receptionist parked a call for me on extension 7001, but when I dialed 7001, just got dead air. I could see in asterisk that the call was indeed parked though, and after calling the person back, he reported he was just hearing the lovely on-hold music. Is there a known issue (and even better, a fix) for this situation? Any other information I can p...

Hello,For the time being I am using the following line to play the original saved message by Asteriskexten => 7001,n,Playback(vm-nobodyavail)Now I am trying to use the other features for Asterisk's voicemail. I have recorded a message, and I can see it saved on the system, but still Asterisk keeps playing the original message... Is there something I can add to let the subscriber plays his recorded message i...

Hello - sorry for my bad english. I will make a list of all sound files on asterisk and i'll record then on professional studio. the french prompts from sineapps sounds bad... sorry for her... tell me if their is many peoples want it ! thank's. en francais: dites moi si ca vaut le coup que j'investissexr dans l'enregistrement des messages en francais. La voix sera la "voix

...tems [SiS] SiS900 10/100 Ethernet (rev 131). IRQ 5. Master Capable. Latency=64. Min Gnt=52.Max Lat=11. I/O at 0xd800 [0xd8ff]. Non-prefetchable 32 bit memory at 0xf3ffc000 [0xf3ffcfff]. Bus 0, device 1, function 3: USB Controller: Silicon Integrated Systems [SiS] 7001 (#2) (rev 7). IRQ 11. Master Capable. Latency=64. Max Lat=80. Non-prefetchable 32 bit memory at 0xf3ffe000 [0xf3ffefff]. Bus 0, device 1, function 2: USB Controller: Silicon Integrated Systems [SiS] 7001 (rev 7). IRQ 11. Master Capable. Latency=64. Max La...

...d(sorry_for_delay) exten => s,4,Goto(s,2) exten => s,5,Hangup ; [insurance] exten => s,1,Background(insurance_thanks) exten => s,2,MusicOnHold(default) exten => s,3,Background(sorry_for_delay) exten => s,4,Goto(s,2) exten => s,5,Hangup if I enter an internal extension (range = 7001 to 7015) then the line hangs up the call. I have just added the [options] section, before it went straight through from s,1 to s,16 and would allow an internal extension to be dialled after s,16. I could/can dial the 444, 555, 666 at any time in the [options] section and the call passes to the cor...

...n't been using R for a while. Many thanks in advance! Cheers Kev Kevin Wang > Senior Advisor, Health and Human Services Practice > Government Advisory Services > > KPMG > 10 Shelley Street > Sydney NSW 2000 Australia > > Tel +61 2 9335 8282 > Fax +61 2 9335 7001 > kevinwang@kpmg.com.au > Protect the environment: think before you print > > [[alternative HTML version deleted]]

Hi, Anyone know where i can find out more info on Reaper/Spawner. Currently, every time i add a new account on my production machine, i have to restart the whole server. After about 150 accounts, this puts a real strain on the server (it takes 3 full minutes before i can access any site on the server). I think reaper/spawner is my answer, but i am havving trouble figuring out how to use it.

...setup" to that. Unless I'm very confused, you > don't really want to see *every* incoming SMTP packet -- just those that > initiate an SMTP conversation. (Note that -- at least in FreeBSD -- the > mail traffic gets logged to /var/log/maillog anyway.) > >> ipfw add 7001 allow log tcp from ${smtp_server} 25 to any > > Again, you may wish to append " setup" to that, for the same reasons. > > In conjunction with the above, you'd likely want to (silently) permit > "established" connections. hadn't dawned on me to this, so:...

....messages=TRUE)' because I want normal behaviour to resume after this particular stop. (Please reply personally as well as to the list, as I'm not subscribed to R-help) Thanks Mark -- Mark Bravington CSIRO Mathematical & Information Sciences Marine Laboratory Castray Esplanade Hobart 7001 TAS ph (+61) 3 6232 5118 fax (+61) 3 6232 5012 mob (+61) 438 315 623

...o MySQL is there an option or anything. Thanks in advance My configuration:: *sip.conf* [general] trasport=udp ;Data format | sample commennt [template01](!) type=friend context=from-internal host=dynamic disallow=all allow=ulaw context=from-internal secret=unsecurepassword [6001](template01) [7001](template01) bindport=6050 *extensions.conf* [from-internal] exten => 7001,1,Dial(SIP/7001,30) exten => 6001,1,Dial(SIP/6001,30) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150917/8223c5a...

...ISTEN Yet, from the outside (either from another PC, or the dom0 itself), I can't connect to the irc ports. I've even disabled iptables on the dom0 altogether, but stil can't connect to it. The only open ports are: PORT STATE SERVICE 80/tcp open http 443/tcp closed https 7001/tcp closed afs3-callback 8000/tcp closed http-alt 8080/tcp closed http-proxy 8081/tcp closed blackice-icecap How do I get this to work? -- Kind Regards Rudi Ahlers -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos-virt...