search for: 6xx

Hello, I wish to play a recorded announcement on reply to 4xx 5xx 6xx messages . According to the status a audio file would be played from asterisk server via ser to the caller How can I configure a such feature ? My configuration: Ser act as an outbound sip proxy . Asterisk a sip media server and registrar. sip agents ---- SER -----Asterisk Harry...

...all from ${CALLERID(num)} to ${DNID} Failed with DialStatus ${DIALSTATUS}" nick at flhsi.com < /dev/null) This works fine, However it's a little lacking. For Instance, Our INTL SIP provider will bounce back SIP status messages if the call is rejected. 503 Service unavailable. 6XX over rate limit. 6XX blocked destination..etc. Anyone have any ideas about how I might capture that and include it in my email. Right now they just all bounce CHANUNAVAIL which is expected. Thanks! Nick Olsen Network Operations (855) FLSPEED x106 -------------- next part ------------...

Dear Gordan, I have a Gigabyte Geforce GTX 560. Is it possible to mod it into a Quadro? If it is possible, please give me the link with the list of steps to mod it into a Quadro. Is it a simple process? Thank you very much. -- Yours sincerely, Singapore Citizen Mr. Teo En Ming (Zhang Enming)

Dear Gordan, I have a Gigabyte Geforce GTX 560. Is it possible to mod it into a Quadro? If it is possible, please give me the link with the list of steps to mod it into a Quadro. Is it a simple process? Thank you very much. -- Yours sincerely, Singapore Citizen Mr. Teo En Ming (Zhang Enming)

...ned", &p->initreq); p->invitestate = INV_TERMINATED; Obviously this doesn't include cases where the URI is not found, the codec is incompatible, etc. More just general failure stuff like executing Hangup() on an unanswered channel. However, 6xxx responses are somewhat religious/political in the SIP sphere... Being that they are "global responses", how could this single Asterisk instance know that this call is unacceptable everywhere/anywhere? From RFC3261: 21.6.2 603 Decline The callee's machine was successfully conta...

...stack -- Executing GotoIf("IAX2/4427-1", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp("IAX2/4427-1", "REALCALLERIDNUM is 4427") in new stack -- Executing Set("IAX2/4427-1", "USEROUTCID=<8xx-6xx-xxxx>") in new stack -- Executing Set("IAX2/4427-1", "EMERGENCYCID=") in new stack -- Executing Set("IAX2/4427-1", "TRUNKOUTCID="Business Name" <5xx-6xx-xxxx>") in new stack -- Executing GotoIf("IAX2/4427-1", &q...

...it possible to add some logic to manipulate the CALLERIDNUM to send back 801 even if the extension is 601 and 901 even if the extension is 701? I have 2 branch offices where users have both Office and Home SIP phones. I want them to share a VM box. Branch1 = 8XX , Home = 7XX Branch2 = 9XX, Home = 6XX Therefore I would like to manipulate the home CALLERIDNUM in both examples. Make sense? Thanks in advance.

Hi, Have looked around for info about this: <http://www.google.com/search?q=Transfer+directly+to+voicemail+site:lists.digium.com> http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMail If we are using 5 digit extensions (10102: 10 for the company, 102 for the extension), where can we put something so that "102*" goes straight to voicemail without waiting while the

...e media path. ? Proxy 1 --> Asterisk--> Proxy 2 ? I want asterisk to anshor media.. ? In extenstions.conf I have an entry to send calls for say 5551000 to proxy 2 and if? I suffix that entry with ||t , asterisk does anchor media.... ? However, it is rejecting the REFER sip method by sending a 6xx message....proxy 2 is trying to send the call back out as a transfer and hence it sends a REFER....how can I make it such that media is anchored as wel as the REFER method is accepted by asterisk.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lis...

Has anyone here ever actually truly successfully gotten a Polycom IP7000 to take a productivity suite license and enabled the bonus features like 4-way calling, recording etc? It ALWAYS works perfectly with ALL of our Soundpoint IP 5/6xx phones, but NEVER for our IP7000s. I just want to know it's POSSIBLE before I keep slogging away at this. Is there a 'bastard_phone=yes' setting that I need to toggle? Also, does anybody know any good therapists with a side-specialty of torn-out hair replacement? :-) Thanks in ad...

...bian sid) with identical configuration. One stands in the server room and is part of the server subnet. the other is in the normal 'user' subnet. I added both server (several times!) to the win2000 domain. the procedure was identical. Both have secret.tbs files of 8192 bytes opposite to the 6xx bytes which they have when they are not joined to the domain. Both 'join procedures' where successfully completed. The samba server in the server subnet is working fine and seems healthy. but the one in the workstation subnet can not verify his secret with wbinfo -t. it returns 'bad s...

..., - { "GeForce GTX 200", "NVA0" }, - { "GeForce GTX 400", "NVC0" }, + { "GeForce 9", "G9x" }, + { "GeForce GTX 2xx/3xx", "GT2xx" }, + { "GeForce GTX 4xx/5xx", "GFxxx" }, + { "GeForce GTX 6xx/7xx", "GKxxx" }, + { "GeForce GTX 9xx", "GMxxx" }, + { "GeForce GTX 10xx", "GPxxx" }, { NULL, NULL} }; -- 2.19.2

Hi All, I have setup 2 asterisk servers using AAH 2.8. I have configured a IAX2 trunk between the 2 servers using the guide on dumbme. Trunk is not using register string and no authentication. In my dial plan I have 7XX numers on server B and 6xx numbers on Server A. Calls from my SNOM phones are ok between the extensions on the 2 servers. In server A I have a eicon 4 port BRI card connected. Calls from outside to extensions on server A works fine. (Inbound routes are configured for "local" extensions). Now how do I configure s...

Hi, i've some trouble with an * installation when the following scenario happen. 1) Inbound call to SIP/xxxxxxxxxxxx ; 2) Call is redirected to ring group 6xx 3) SIP extension 1xx answer. 4) caller want to speak with john doe on his mobile 5) assistant put caller on hold 6) assistant start a call to john doe mobile using a php script (AMI - Originate with custom context to force outbound trunk) 7) if john doe want to speak with caller assistant bridge th...

Ok here is a question that has gotten me stumped. I have an Asterisk system up and running. I need to connect it via the Internet to a Sip Cisco system. This is what they have. I have there IP address's and login. X-lite is able to connect to them and make a call! So I have the name right! CISCO router model: 2621 VoIP module: NM-HDA-4FXS I have done Google lookup and at the Wiki about

..."TearFree" Bump version for 7.8.0 release Qiang Yu (3): Remove RR_Capability_SinkOutput for GPU without CRTC Fix radeon_mode_hotplug crash on multi GPU platform. DRI2: Fix radeon_dri2_exchange_buffers width/height copy'n'paste error Tan Hu (2): EXA/6xx/7xx: fast solid pixmap support EXA/6xx/7xx: accelerate PictOpOver with component alpha Tom St Denis (1): dri3: Return NULL from radeon_dri3_pixmap_from_fd if calloc fails. git tag: xf86-video-ati-7.8.0 https://xorg.freedesktop.org/archive/individual/driver/xf86-video-ati-7.8.0.tar.bz...

Hi, I ve had some problems printing from NT, I had samba 2.0.3 and every time at the end of document there is a blank paper and I don't know how I can stop this ??? Regards, and thanks in advance for any help Amin.

....D*Y-R%GMA42>QKDK)#C6XMASYJ27"R35[+7GRXZ^*K?R>;1MV]2$`&] M+(BRS8[9'^)V]T,ODBU\A^Q9L]-9$A5F,[O(]0:7P]Y'L_O]8N?-;JMC9O-Y M*?5U7YRL/;$\'-+@Z4,ME1K%ZI&6T-*A+N161(H-H][0ZT?5NJY5ZJOG62X7 MX2`+T2#.$PJ&4QM1CR/.+"^<4HX9"0/?"PF^HWR"K1F?Y&!S,YLC,(:#6XX. M`T589^AP&HX5M+3*R-\S$@('O"0\H-=1I`E&7&K=N"19$9E#6/[=-C^DKA>, M"Q'JV>L&Q)='[FY`F3UYP<+.&]W1EOJ4+>K=4DYV,!`2SPF=^;QP:_D]BEDR M<WL_=?&,N4KTZ="0@X.<WA+,J#=.V/PPX/YGXBG1#B@'+PO_.R]M:'-8]+GG MW-YRD)'CT4F"7YL[V&qu...

I was afraid someone would say that :-). I am currently at 3.10.0-862.3.2 and downgrading to a 3.10.0-6xx kernel breaks some of the other packages. Is the mentioned bug documented somewhere? Cheers frank On 07/06/18 06:22, James Peltier wrote: > There was a kernel bug that affected all NFSv4 traffic that we ran into > and we had to downgrade the kernel to and older version > (3.10.0-693...

...:- -- Accepting overlap call from '123' to '6' on channel 0/31, span 1 -- Starting simple switch on 'Zap/31-1' -- Hungup 'Zap/31-1' Primary Rate E1 30 trunks connecting between Asterisk and TDA200 Pansonic TDA200 has 1XX extensions Asterisk is setup with 6XX extensions If Asterisk calls a 1XX its not an issue, when 1XX calls Asterisk it looks like the phone system is dialing the digitals individually instead of at once so Asterisk is receiving the first 6 going I don't know 6 before it receives the rest of the digits from the TDA. Any clues as to...