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the CNAM info in the Q.931 call setup message. I've tried all permutations of switchtype (dms100 & national) and facilityenable that I can think of, but I still don't see CNAM coming out the other side. Telco confirms that "Name Out" is enabled on our PRI. Any pointers on what I'm missing, and/or how to debug further? zapata.conf: --- [channels] context=default

Dial(UniCall/g1|300|) Where is the number you want to reach? I'd expect to see Dial(Unicall/g1/1234567890|300) To reach number 1234567890 - Mois=E9s Silva On Jan 30, 2008 1:21 PM, Roger C. Beraldi Martins <rogerberaldi at gmail.com> wrote: > Dears, > > After weeks trying to contact support of my telecom about 'Seize Ack' > because that is not returned, was a

Hi Folks, I'm trying to send CallerID Name information out to the PSTN via a PRI with Bell Canada with no success. With inbound calls (originating from the PSTN) CNAM is received successfully, and we've not had any similar problems with other Telco PRIs, so I'm stumped.

And I'll resend this one too. Silly scalix. --Rob -----Original Message----- From: Rob Thomas Sent: Thursday, 22 June 2006 12:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] zapata.conf: recent changes? Looks like you've stopped compiling libpri. All those options that are being ignored, are being ignored because they're

Hi Group, I am currently facing a dead end and any help will be much appreciated. I have an a104d installed in an asterisk box, two of which is configured on ISDN pri. One is facing pstn and the other one is facing a hipath 300e Siemens. I am getting one way audio when a local on the hipath tries to make a pstn call but no issue on incoming calls from pstn going to the hipath locals. local

Hi, after a few of upgrades, I noticed these messages in full debug log: Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring switchtype Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring pridialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring prilocaldialplan Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring overlapdial Jun 21 12:58:11 WARNING[27273] chan_zap.c: Ignoring

Hey, Thanks for the input Andrew. I did all you suggested but noticed that when I did the loopback test, the output *was not* there as you mentioned ("I'm set to pri_net, but the other side thinks it is pri_net!"). In fact, the same message as before kept repeating every second or so: >> Unnumbered frame: >> SAPI: 00 C/R: 0 EA: 0 >> TEI: 000 EA: 1

Dear Customer Support, i connected the asterisk to a e1 interface of our hipath4000. outgoing calls from a sip peer of my asterisk to an up0 telephone which iss connected to the hipath4000 are working. If you want to dial from an up0 device to the e1 interface where asterisk is connected to, you have to use the prefix 83. But when you enter the 3rd cipher this error appears at the cli

Although in a bugtracker posting with a patch from over two years ago, Matt Fredrickson from Digium says that it works with 5ESS under Asterisk 1.2.X: http://bugs.digium.com/view.php?id=3554 There are also bounties and claims of this feature working on NI2 protocol(although no patches posted) on the voip-info.org Wiki:

I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card. I can't make any connection over the T1. From CLI: ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling method 'pri_cpe' at line 37. cat chan_dahdi.conf cat chan_dahdi.conf [trunkgroups] [channels] language=en ;internationalprefix = 00 ;nationalprefix = 0 context=from-pstn switchtype=national

I have seen conflicting references in regards to the Digium Wildcard TE411P echo settings in zapata.conf. Does anyone have the official word on this? Should echo cancel be enabled in zapata.conf if the card has built in EC? If so, should a particular EC method be compiled into the zaptel build? My reference, which has echo: My zaptel is 1.2.5 context=from-pstn switchtype=national

XO CAN supply callerid NAME on a NI2 PRI connection. We have three of them and they work great. Its takes a little doing to get to someone at XO that knows what they are doing but XO does have some VERY good tech support people that know how to get things done. It just takes a bit of work to find them. Outgoing CNAM is a different beast however. They can't take it via IE. You need to get

I have not been able to get two B-channel transfer to work on DMS100 PRI. I consistently get the following errors: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ROSE RETURN ERROR: [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: OPERATION: RLT_OPERATION_IND [Nov 6 11:12:49] ERROR[2774]: chan_zap.c:8178 zt_pri_error: ERROR: RLT Not Allowed I have tried on two

I have spent some time this morning trying to add an Astribank to our current Asterisk, but it failed, so I just removed the hardware, restore the config files to the original setup and started asterisk.; I could see that no Zap channels are started so I did load chan_zap.so: pbx*CLI> module load chan_zap.so [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application: Already have an

Dear All The setup is te110p with an 8 channels PRI to make and receive all calls. SIP phones throughout the company. TDM400p with 4 FXS modules to send/receive faxes and make credit card transactions. I have an analogue phone on the tdm400p for testing. I can receive calls to the exten. There is a dialing tone. However, when I try to make a call I get a busy signal. Asterisk stated busy then

Jan 19 10:00:29 VERBOSE [7177] logger.c: -- Executing [1000 at ext-meetme:7] Read("DAHDI/2-1", "PIN|enter-conf-pin-number||||") in new stack Jan 19 10:00:29 VERBOSE [7177] logger.c: -- Playing 'enter-conf-pin-number' (language 'en') Jan 19 10:00:43 VERBOSE [7177] logger.c: -- USER ENTERED 'THE PIN NUMBER' Jan 19 10:00:43 VERBOSE [7177] logger.c: --

We have a DMS100 that does not have PRI. So we're using a channelized T1 using WU-LAW, ESF and B8ZS coming from the DMS100 that's plugged into a Tormenta2 Quad T1 Card on my Asterisk Box running Debian 3.01(woody) with Kernel 2.4.22. The Link is up but according to the DMS100, Channel_1 goes into RMB (Remote Manual Block) and Channel_2 goes into LO (Lock Out). I've been through all

2014-02-06 11:09 GMT+01:00 giovanni.v <iax at keybits.org>: > Il 05/02/2014 8.42, Olivier ha scritto: > > channel then it depends upon what you have the priindication option >> set to. With >> priindication=outofband then a busy cause code is sent to the >> network and the call >> is hung up. With priindication=inband then a busy tone

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > brett-asterisk@worldcall.net > Sent: Tuesday, November 30, 2004 2:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] National (US) callerid name > resolution for yourasterisk box >