similar to: no incoming pstn ring tone

Hi, I have an asterisk installation that has been happily working in production for some time (E100P and UK BT ISDN30). Recently I upgraded to HEAD-07/29/04. Now, incoming callers don't hear ringing while calling in. As far as I can tell, my config files haven't changed from what was working before. Can anyone please help before my boss shoots me? JC zaptel.conf

Hi, is there any mean to get ringing tones for the E100P, according the signals from the telco? When I use the r-option in extensions.conf, I have too early ringing tones, i.e. when busy, I have ringing tones for a while and afterwords the busy tone. If I don't use the r-option I have no tone at all. Is there anything similar the b-option for chan_capi? Roger.

Hi, I'm calling asterisk with a swedish PSTN-phone line with caller presentation (DTMF) activated. I'm using asterisk 1.4.20.1 and cannot upgrade unfortunately, so I have to stay with this release. I use a TDM800P 8 channel PSTN card working as answering phones (I connect a phoneline with carrier signal to my TDM-card). Using zaptel-1.4.12.1. I verified that the DTMF tones of the number

Hello, I am using an incoming iax provider to bring calls to my server. When an incoming call comes in, the calling party does not hear a ring tone. I figured that this was no big deal, and that I just needed to enable the "r" flag to dial. This has not fixed the situation though. Just to try to make sure the line was being picked up properly, I tried the following: [inbound] exten

Hello All, one of the extensions fall into a loop, I don't know how to hangup that channel -- Executing [i at autoatten:2] Goto("Local/100 at sipphones-000001b2;2", "s,2") in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200 at sipphones-000001b2;2 -- Executing [i at autoatten:1]

Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104 at sipphones:1] MeetMe("SIP/101-00000813", "104") in new stack == Parsing

Hi All, I recently upgraded from a very old stable to HEAD. For some reason, incoming callers don't hear ring tones when calling in. Everything else is working fine. Where should I look for a fix? ISDN --> X100P --> asterisk --> sipphones. Thanks Johan

I've been asked to recommend a solution for a one-E1-port PSTN gateway supporting SIP. I've never set up a Cisco 5300 or equivalent, but I know they work. I use the Asterisk software in a couple of places and would like to use the E100P. My question is whether anyone out there has any installations using this and what their opinion is about it (does it work? how's the audio quality?

Hi, I have a problem with TDM2400P card. The card is detected ok, I can make a call but only with pulse dialing (not tone dialing) without hear sounds from the headset. When I receive a call, I can to establish a communication, but without hear sounds from the headset. When I dial any phone key, I can hear dtmf tone. I'm using Elastix 1.5.2. These are my configuration files:

Few weeks back local telco introduced option of custom ring tones. I am not talking about your phone ring tones but about ring tones you hear in your headset while phone is ringing on the other end. If I understand correctly, ringing tone is generated localy on asterisk if you are connected to phone network with E1/T1 connection. Which means that instead of regular beep-beep tone we could send

I need to process a number of lines of code in the dialplan before answering a call. Can standard ring back tones be played to the caller while this is happening prior to answering the call. Which commands would facilitate this? Thanks in Advance, Vic

Hi, I had asterisk installed, ISDN-adapter, some x-lite software-phones and I can call betweens the softphone- and 'normal' phones during the ISDN-card. 2 questions now 1) Is it posible to create policies, so that some SIP-users can dial ALL numbers, and some SIP-users not are allowed to dial eg. 900xxx-numbers, 30xxxxxx (mobilphones), 40xxxx(long distance) and if possible on time

Hi everybody,, I need to block incoming collect calls to my Asterisk box but I could not find out where to do that. Went to zaptel.h but I did not see any timing which can be applied to collect calls. Does anybody knows if I can set this up in Asterisk? I'm using an E100P connected to the PSTN and a T100P connected to a Zhone 100. Version: Asterisk CVS-05/30/04-16:28:04 thank you Oz

Hi- Perhaps someone with an E100P in hand can answer this: I just received an E100P from Digium (I normally buy quad boards) I noticed that the circuit board says "T100P" on it, and I assume that the T100P and E100P both use the same circuit board. Can someone please confirm that their E100P says "T100P" on the artwork? Thanks Scott Scott M. Stingel Emerging Voice

hi guys, I have a little problem maybe you can help ... I have an asterisk setup, with an E100P, and an ISDN-PRI 30 channel line from the telco going into it .. the E1 line is OK, because plugged into a Lucent Portmaster 4 it works OK .. plugged into the asterisk box I just get an engaged tone, and asterisk posts this message on screen : WARNING[1167272000]: File chan_zap.c, Line 5275

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

Hello ml, i need some help on my zaptel configuration. My E100P only shows some YELLOW / RED alarm when I load the wct1xxp module and do a cat /proc/zaptel/1 Span 1: WCT1/0 "Digium Wildcard E100P E1/PRA Card 0" HDB3/CCS YELLOW RED ... .. . My /etc/zaptel.conf is: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=nl defaultzone=nl I tried zaptel-1.0RC2 and the latest CVS

Just cross pairs 1-2 and 4-5 1 <--> 4 2 <--> 5 Never done an E1, but I think it's the same for a T1 4 wire cable. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: bam [mailto:bam@cqm.co.uk] Sent: Monday, December 29, 2003 9:12 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] E100P pinouts

Hi ! I have a E100P and I would like to receive incoming calls on dedicated channels only. Is it possible to answer an incoming call request on channel 2-30 from the Telco with something like 'busy, use channel 1 instead' ? If this is possible, how could it be implemented / configured ? Br / Jan Terje