similar to: SIP 603 response when call is not answered

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

Hi To troubleshoot FreeBSD panics triggered by ISDN load on an asterisk system, we are looking to buy an ISDN call generator/simulator device. The minimum requirements include: - Not too expensive - PRI support (BRI support is a plus) - CCS+CRC4 farming + HDB3 coding - EuroISDN (DSS1) support. - A minimum of 4 ports (120 channels/concurrent calls) - Compatibility with Digium cards. - DUT in TE

On 2016-08-29 12:28, Eric Klein wrote: > Hi Hooman, > > What you probably want is a PRI PBX running Asterisk. > > You should either plan to build your own (with the cards you need) or get one of the low cost options: > > * Allo.com has their Mega PBX with 1 PPR port (http://allo.com/megapbx-line.html) > * Pika Tech has the Warp PBX with BRI

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk,

hi all, i have an asterisk install with a digium 4 port fxo card and cisco 7960 sip phones -- running on a compaq Pentium III (Coppermine) at 800Mhz 256KB cache and 1GB of ram. when a call comes in on zap/1-1 for example, the delay between when zap sees the line going to ring state, and when the desktop telephone rings can be as long as 7000 milliseconds (or about 3 or 4 rings on an ear piece).

In Asterisk 1.8, I used the following line in extensions.conf to allow me to pass "*82" in front of a dialed number, to disable the callerid block that's normally on that POTS line: ; disable callerid block exten => _*82.,1,Dial(${POTS}/${EXTEN}) But this seems to have stopped working when I upgraded to Asterisk 11.7. I get the following debug output, with a "no

I've discovered that the status of a SIP device doesn't get passed as in-use when on an outbound call. Viewing the debug log the status is always passed as 'not in use' when on the outbound call. The sip_devicestate function doesn't appear to check the user object at all. The devices are configured as friends in sip.conf. Being both a peer and a user, the device is

hi@all I have the following problem: With asterisk 1.09 the grandstream's registers fine with both ports, with version 1.2.1 (the newest port on freebsd) I get "Unauthorized" SIP messages from the 2nd port. The ports are configured identically, the only difference is the sip and rtp port. On the first port the sip port is 5060 on the second 5062. The rtp on the first 5004 on the

Hi, I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers. I installed the new driver (0.3.1-rc30) on our pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]:

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

Could of requests: 1) Is there any example or writeup on the difficulties of extending reference classes across packages? Just so I can fully understand the issues. 2) In what sorts of situations does the performance of reference classes cause problems? Sure, it's an order of magnitude slower than constructing a simple environment, but those timings are in microseconds, so one would need a

Hi, Has anybody come across a situation where they were unable to register with Asterisk through a SIP stateless proxy server? I'm getting an error: "403 Authentication user name does not match account name" As far as I can tell the requests reaching Asterisk with and without the proxy are identical excepting the IP address the REGISTER request is coming from and the Via header

Hello, CVS 07/31/03. Test with 130+ PSTN-to-SIP calls. Asterisk gets locked ... grep -e "Error" -e "eventually" p-console chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got it eventually... chan_sip.c line 1453 (sip_alloc): Error obtaining mutex: Device or resource busy chan_sip.c line 1453 (sip_alloc): Got

hi, I'm working with asterisk on a project and I found a problem with cdr_odbc. As we know, after answering each call a cdr event is raised which is saved in cdr_csv and cdr_odbc. but here my point is on cdr_odbc. some information, including start_time and end_time is given by cdr event but the problem is that these two information(start_time and end_time) is not getting save in cdr_odbc. I

Hello everyone, I can not get the name of the recoding file of agents calls. I set agents.conf as following: ; Enable recording calls addressed to agents. It's turned off by default. recordagentcalls=yes ; ; The format to be used to record the calls (wav, gsm, wav49) ; By default its "wav". ;recordformat=gsm ; ; Insert into CDR userfield a name of the the created recording ; By

[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>

Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the

I am using dovecot lmtp root at messagerie[10.10.10.19] ~ # grep virtual_transport /etc/postfix/main.cf # transport_maps = hash:/var/lib/mailman/data/transport-mailman, proxy:mysql:/etc/postfix/mysql-virtual_transports.cf # virtual_transport = maildrop virtual_transport = lmtp:unix:private/dovecot-lmtp root at messagerie[10.10.10.19] ~ # On Thursday, February 9, 2017 7:54 PM, WJCarpenter