search for: sandesh440

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 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 d...

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:

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

...X,n,ExecIf($["${PRIVACY}" = "id"]|SetCallerPres|prohib) This makes the calls with privay ON sent as anonymous at the other end. One more thing is to make sure you enable usecallingpres=yes in chan_dahdi.conf. Thank you Sandesh On Fri, Mar 5, 2010 at 11:18 AM, das sandesh <sandesh440 at gmail.com> wrote: > Hi All, > > We have two servers, one server (SIP asterisk server) sending calls to the > second server(has PRI) which goes our through the PRI's (using TE 412p). > When the pprivacy is enabled: P-Asserted-Identity Header, privacy "id" are >...

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, 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, 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 All, I have a senario where we have multiple locations and all have the ability to call using 1NXXXXXXXXX pattern, so we have created multiple contexts so the outbound goes fine, but while transfer occurs (after picking the inbound call and transfer), it uses the first 1Nxxxxxxxxx priority patterned context, like if the 3rd location is making a transfer, but 1st location have the priority

Hi All, I was actually trying to use the dialplan application that uses 'Dial' and when the: Dial(SIP/XXXXXXXXXX at xxxx|20|) command is executed and the destination number rings for 20 sec after which I receive as "503 Service Unavailable", but not "480 Temporarily unavailable". Dial(SIP/XXXXXXXXXX at xxxx|20|) exten => XXXXXX,n,NoOp(Dialstatus:${DIALSTATUS})

Hi All, We have two servers, one server (SIP asterisk server) sending calls to the second server(has PRI) which goes our through the PRI's (using TE 412p). When the pprivacy is enabled: P-Asserted-Identity Header, privacy "id" are sent in the header of SIP invite packet to the second server, how can we identify this privacy and block the callerid as the call goes to the second

Hi.. We are facing a problem that is making the channel to be stuck. we are using asterisk 1.4.22.1 version, and we have a direct sip trunk...We have 2 queues and one has 2 agents and the other 5 agents, from last week the second queue's channel is getting stuck, it happened 3 times till now and the problem is calls come into the queue and just the calls will be in the queue and will not ring

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 Leif, Thanks for the information. I checked the /tmp/ folder and there was core #### files and I tried to back trace it but it was not showing the cause of that crash, but anyhow, I upgraded our Asterisk system to 1.4.22.1 and from past few days its going on fine. I have also researched and found that version 1.4.17/18.1 had the issue of channel stuck up as well as random asterisk crashes.

Hi All, We are using group paging and our asterisk version: 1.4.22.1, but when ever any one page to the whole group (28 extensions), the calls which are on hold on those extensions will be dropped, is there any other way to have this feature or to go with Overhead paging. Currently this has become a serious problem, can anyone through some light on this group paging senario? Thank you very much

Hi, Asterisk got stopped this morning after 20 minutes and phones went to 'No Service' and then got started automatically after 20 min, as I could see in the full log that asterisk got started at so and so time: [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log [Feb 10 08:29:31] VERBOSE[31013] logger.c: Asterisk Dynamic Loader Starting:

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 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