similar to: Regarding - P-Asserted identity

Hi All, I got this figured out, when the privacy is ON at the other end of the server and when we get the Invite message to the server connected to PRI's, just take the details from the invite message in the Dial plan and send the calls as anonymous: exten => _1NXXXXXXXXX,n,Set(PRIVACY=${SIP_HEADER(Privacy)}) exten => _1NXXXXXXXXX,n,ExecIf($["${PRIVACY}" =

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk:

device somewhere in your communication path, and since voice is picked up as DTMF, some device is also set to listen for inband DTMF. What is the origination source of incoming calls to your system? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-08 4:24 PM, "das sandesh" <sandesh440 at gmail.com> wrote: Hi, We have few systems with asterisk 1.4.22.1 and we use sip

Hi, I was trying to use 2 of asterisk servers and interconnected, one of them as a peer to other sever (configured in sip.conf), so all the calls to server 1 will just be passed to server 2 (has PRI Card, TE 412P, only one PRI connected), i was sending calls to server 1 and that would send to server 2 and then dial out using Dahdi, but the problem that i got was the hangup cause codes, i was not

Hi All, We have built an asterisk server (asterisk - 1.4.26.2) where there would be around 322 concurrent calls going on, but I can see that full log grows rapidly, in one day it reaches to around 10-15 GB if I turn on the sip debug and its tedious even by using any commands to get the required call from the log if there is any problem. Is there any way of splitting the full log into parts

Hi, We have few systems with asterisk 1.4.22.1 and we use sip trunking for them not PRI's, one of our system is giving a problem of dtmf (rfc2833), like when we dial the number that have IVR and enter the extension or access code, it some time takes it and some times does'nt recognize the digits dialled. We also tried auto and info for dtmf but could not get the dtmf to work reliably, can

Hi All, We are trying to implement a DS3 capacity calls (672 concurrent calls) using asterisk server. I wanted to ask are there any compatible DS3 cards with asterisk? I tried searching a lot but could find DS3000P from digium but unable to get this product. Does anybody have any idea of having any DS3 card in asterisk box so as to handle around 600 calls? Thanks Sandesh -------------- next part

Hi, I tried getting our server setup for 400-500 simultaneous calls, calls were going through properly but at around 200-250 calls, mysql (connect ...) statement was taking at least 5-10 sec to connect to the database. I optimized all possible parameters in my.cnf: max_connection=1000 wait_timeout=60 query_cache_type=1 query_cache_limit=4M query_cache_size=512M interactive_timeout=120

Hi We are facing a problem for orphaned parked calls, we have the following config: asterisk -1.4.22.1 dahdi-linux-complete-2.2.0.2+2.2.0 and when we get an incoming call and after it gets parked, after some set time (here its 2 min), it goes back to the operator, but the problem is that randomly it tries to call SIP/5060 instead of SIP/2200 (where 2200 is the extension number of the operator)

Hi, We are experiencing this issue of redial dtmf tones generated randomly in the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one FXS is used for Fax and rest are empy) connected to the netgear switch and all the phones are connected to this switch and there are no non sip devices in the

Hi All, I have a TE121P card installed and since connected it to the PRI I keep getting the Current Alarm as continually changing from Blue Alarm/Recovering and Recovering. The config I have is: /etc/dahdi/system.conf bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone = cn defaultzone = cn /etc/asterisk/chan_dahdi.cfg [channels] Context=telco

Hi all, We have our system hosted publicly and 4 phones are connected remotely at employee's home, and when they try to do a assisted transfer to one of the employee at the main office, the call is lost. For ex: person A calls person B, person B calls person C for assisted transfer, and as soon as person B hits transfer button again to transfer person A to C, the call is lost. But in the

> > Funny. The first thing I always do after a reboot is call in from my > cell to make sure things work. But last night I rebooted and immediately > tried dialing out (with a TDM842B) and got: > > WARNING[2720]: app_dial.c:1721 dial_exec_full: Unable to create channel of > type 'DAHDI' (cause 34 - Circuit/channel congestion) > > Restarting all of Asterisk or

Hi all, A few days ago I found out with help of some of you guys how to set CLIR. (Calling line identification restriction) My first idea was to use the keypad protocol to set the CLIR with dialing *31* before the number but this was not possible. So thanks to Damon Estep I got it to work with executing 'SetCallerPres(prohib)' before the dial command. This works perfectly! But now

Hi All, Today I faced a problem with channels getting stuck. We use asterisk 1.4.18.1, and there were 2 extensions (channels) that got stuck. When I try to do "soft hangup <channel>", it says "Requested for soft hangup" for that channel, but if we go and check once again those channels are still stuck. Also even after asterisk restart it did'nt go, finally we had to

Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI> == Using SIP RTP CoS mark 5 -- Executing [100 at sip:1]

Does anybody know if SetCallerPres works on calls via SIP through a Cisco gateway? We are trying to mark outbound calls as anonymous and we set it to prohib, but calls still show outbound callerid. We are SIP from * to the Cisco gateway and then PRI outbound. If we strip the callerid num, then the first number on the PRI gets added as teh callerid, so we can't do that. We need to make

Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! -------------- next part

I have an issue where a particular dialplan works but another doesn't. I'm not sure why. To me they look identical and it has me stumped. This works: [to-test] exten => _X., 1, SetCallerPres(allowed) exten => _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb) exten => _X., 3, Ringing exten => _X., 4, Dial(SIP/9330 at a-test,20,ro) exten => _X., 5,

Hi, The case I'm working on is : - a call comes from PSTN to a given extension (say 122) - a user picks the call up (dialing *8122) from another extension (say 240) using a SIP hardphone - the hardphone (he one with 240 extension) displays the dialed string (here *8122) instead of original caller-id. This is logical but I would like to change this default behaviour so that original