similar to: DTMF with grandstream phones

I have a Cisco 7940 when you call in from outside and dial the Cisco phone extension I get this Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[18451]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3)

Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX-XXXX 3- Anything else that should be restricted if one was to restrict all calls to US 48 only I have found many list but it's tough looking at the entire list of area codes and pulling out each of them

I am seeing some curious behaviour with a 1.2.32 system, which I do not understand and so can't work out how to fix it. I have a PRI routed to context default. Here is the complete default context: [default] exten => _9X.,1,Dial(IAX2/m1peer/${EXTEN:1}) exten => _20XX,1,Dial(IAX2/sipeer/${EXTEN}) exten => _X.,1,Dial(IAX2/m1peer/${EXTEN}) exten =>

Are there any sip providers out there providing full business telephone service. Not just single line/residential service like I have seen with vonage etc. For example take a company currently using a legacy pbx connected to the PSTN with a PRI. I would like to replace this setup with a data T1, an asterisk box, and some SIP Phones then pass all calls (local and long distance) directly

Is it possible to give a false ringbakc on asterisk ? -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or

Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: ***

I switched PRI vendors recently, and one of my questions was "do you provide caller ID name in addition to number?" AT&T Local did not, But XO communications said they did. Before I call to complain, is there an setting to turn this on in asterisk? I want to make sure that I have my side covered before I call XO. My current zaptel.conf is: context=from-pstn switchtype=national

Hello, I just got off the phone with my PRI provider, and was told that I am not sending an expected message when I reject a call with a Cause Code for Unassigned(1) and Congestion (42). Busy works fine though... Anyhow, they are seeing the RELEASE COMPLETE message with cause code 1, however the tech told me they expect a PROGRESS indicator with a value between 1 and 10. Any ideas on how

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the

Hi, I have an asterisk installation with 2 E1 cards Software version is Asterisk 1.2.6 Libpri 1.2.2 Zaptel 1.2.5 I'm having problem with fax transmission, let me explain better my setup: My fist TE110P E1 card is connected to the telco line the second TE110P E1 one to an Nexspan PBX so the server is basically sitting between the line, and the pbx. every call coming from the line is

I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through Asterisk, and based on them, build the destination PSTN number). My problem is that Dial send the DTMF's to the SIP/gateway user at a rate

Hello, One quick question. Does anyone has experience implementing unified messaging (UM) using Asterisk. Does Asterisk has support for UM ? Thanks, Tarun ___________________________________________________ Medicine meets Marketing; Dr. Swati Weds Jayaram. Rediff Matchmaker strikes another interesting match !! Visit http://matchmaker.rediff.com?2

This is BAAAAAAAD! Now even SIP get's "tainted"... http://slashdot.org/articles/04/09/16/1317247.shtml?tid=95 _________________________________________________________________ Surf the net and talk on the phone with Xtra JetStream @ http://xtra.co.nz/jetstream

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

hi, well, most of the things work right now due to the help of peter svensson, but after heavy use of our ericsson BP250 today several problems appeared. i split into several mails as they are seperate problems. * i can't signal Busy to the calling party. asterisk receives busy from the ericsson PBX but does not forward this to the external caller. i tried with exten =>

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

I've been beating my head a bit against the 1.0.6 Debian builds of Asterisk, using an E100P (E1, single span) board. In machines I've built in the past (back in 1.0.0 days), config I'm using and that card and 1.0.0 driver combo worked fine. ztcfg reports no problems: SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 31 channels configured. And zttool sees the card, and

Hello, i have succesfullu setup asterisk with Sangoma E1 card, evrything works well but i don't know how to pass indications from telco switch to the user - when users call bad number telco switch shuld talk "unallocated number" but its only send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients? My /etc/zaptel.conf: span=1,0,0,CCS,HDB3,CRC4 dchan=16

so here it is, the problem that's been nagging me for the past 2 days: connected a box to my telco's NTBA <-> zap/asterisk. which works: box:/etc/asterisk# cat /proc/zaptel/1 Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" HDB3/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In