search for: 10126

...35396 INVITE *User-Agent: Grandstream GXP2010 1.2.1.4* Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE Content-Type: application/sdp Content-Length: 250 v=0 o=user 8000 8001 IN IP4 192.168.1.102 s=SIP Call c=IN IP4 192.168.1.102 t=0 0 m=audio 10126 RTP/AVP 2 8 101 a=sendrecv *a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000* a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> [Aug 2 13:56:57] --- (14 headers 12 lines) --- [Aug 2 13:56:57] Sending to 192.168.1.102 : 5062 (NAT) [Aug 2 13:56:57] Using INVITE reques...

...denuncia dei redditi, qualsiasi costo relativo a servizi acquistati presso di noi. Per qualsiasi informazione aggiuntiva non esiti a contattarmi al mio indirizzo di posta elettronica. Scusandomi ancora per il disturbo arrecato Le porgo distinti saluti. Boeris Flavio DISCOVERYWORLD C.so Dante 41 10126 Torino mailto:fboeris at discoveryworld.it -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or "[un]subscribe" (in the "body", not...

...ing back to context 'default' Do you have some idea to achieve this kind of result ? Maybe I can use ChannelRedirect from Asterisk 1.4 ? Cna you give me a hint on that ? Tnx in advance Regards Edoardo Serra WeBRainstorm S.r.l. -- Ing. Edoardo Serra WeBRainstorm S.r.l. Via Pio Fo? 83/C 10126 - Torino Tel: +39 011 678 100 Fax: +39 011 678 275

Hello all, I currently have all outgoing calls set to mask the caller id so it will always appear to be coming from our main number. The problem I'm having though, is with both the call detail in mysql and with the automon (recording) feature. It shows the originating number as the number I masked it to, rather than the actual person calling. How can I go about having both the destination see

I am looking for a way to automatically close a meetme conference when either a user hangs up or through an agi call? Some method that would automatically terminate the meetme. Is there a way to do that? Jerry

All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be ideal. Any suggestions? - sf

Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, When dialling a SIP phone which is already in a call the caller hears a "regular" ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi:

Within the RJS template is there a way to specify when the visual_effect should occur? I''m trying not to repeat the :before and :success blind_down visual_effect statements throughout my views. It seems like the rjs templates can only take visual effects statements in the form of page.visual_effect :blind_up, ''divname''. I can''t figure out how to add

I would like to know if anyone here knows the answer to the following question I need to implement the following conferencing feature for my agents. 1. Agent receives call from caller 2. Agent conferences a verification service 3. After finishing the verification, agent needs to drop third party (Verification service) and continue on the line with caller. My problem

...rs:*:513: Domain Guests:*:514: Domain Computers:*:515: Administrators:*:544: Account Operators:*:548: Print Operators:*:550: Backup Operators:*:551: Replicators:*:552: sistemas:*:1002: LAPAZ\domain admins:x:10125:LAPAZ\root LAPAZ\domain users:x:10124:LAPAZ\compu1,LAPAZ\foobar$ LAPAZ\domain guests:x:10126:LAPAZ\nobody LAPAZ\domain computers:x:10127:LAPAZ\serverlpz$,LAPAZ\compu1$ LAPAZ\seccion:x:10128: on serverlpz serverlpz:~ # getent passwd at:x:25:25:Batch jobs daemon:/var/spool/atjobs:/bin/bash .. root:x:0:0:Netbios Domain Administrator:/home/root:/bin/false nobody:x:999:514:nobody:/dev/null:/b...

hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because

Hi, I've just installed Samba 2.0.3 and now I'm getting following error: PAM_pwdb[306]: 1 authentication failure; (uid=0) -> andreas for samba service I can connect to all shares and printers, but get the error. I suppose it has something to do with the passwords (I'm trying to sync Samba and Linux passwords). Any ideas? /Andreas

Hi, I am seeing the following (with Dovecot 1.0 rc10): Oct 18 07:35:58 dovecot: IMAP(user): UIDs broken with partial sync in mbox file /var/mail/user Oct 18 07:35:58 dovecot: IMAP(user): mbox sync: UID inserted in the middle of mailbox /var/mail/user (15314 > 9604, seq=5, idx_msgs=224) Oct 18 07:35:59 dovecot: IMAP(user): file mbox-sync-rewrite.c: line 405 (mbox_sync_read_and_move): assertion

Help please- I am running 3.3.1 on Centos using a 10GB network. I get reasonable write speeds, although I think they could be faster. But my read speeds are REALLY slow. Executive summary: On gluster client- Writes average about 700-800MB/s Reads average about 70-80MB/s On server- Writes average about 1-1.5GB/s Reads average about 2-3GB/s Any thoughts? Here are some additional details: