similar to: Shorten the recognition time of rings on Wildcard X100P

Hi! I have a problem with one box running asterisk, one pots line and an X100P. Almost every night the phones give 2-3 rings and then stop. There are no actual incoming calls, I verified by putting a device that lists the incoming telephone numbers parallell to the X100p and it doesn't list any calls. This is the output on the console for a real incoming call: ? == Spawn extension

I am looking for a supplier of Digium hardware in Canada. Any suggetions? Thanks, Adi

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

Hi, I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm having problems with Caller ID. I have run clidtest, and it seems happy enough, returning:- server clidtest # ./clidtest /dev/zap/1 Number: 0412222222, Name: MOBILE (that number's fake.) However, I'm not getting the caller ID passed through with *. Sometimes I get a "failed checksum" like

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware

I admit that this might be some very basic question... How do I obtain Asterisk 1.0.3 from CVS? Does '-r v1-0' get me 1.0 or 1.0.3? Thanks, Adi

hi, when i call asterisk (on x100p) i got this : CLI> -- Starting simple switch on 'Zap/7-1' Jul 13 15:03:34 ERROR[311316]: callerid.c:192 callerid_feed: fsk_serie made mylen < 0 (-9) Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4735 ss_thread: CallerID feed failed: Success Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4777 ss_thread: CallerID returned with error on channel

I have an Asterisk server running todays CVS (updated it just in case that was the problem). It has 3 X100P cards. The first two lines I use as my normal work lines and the third is my fax line which I use with SpanDSP. I run Fedora Core 4. I have a problem that the third X100P does not answer the call. From the console I can see that there is an incoming call with the following: --

Hi Folks, Prior to upgrading my Zaptel sources everything was working fine. I have a X100P connected to my analogue line. The handset port of the X100P is connected to my desk phone's line 2 input. When the analogue line rings I see the CID on my line 2 but not from Asterisk on line 1 via the Cicso ATA. This used to work fine until I upgraded the sources. I get this when watching the

I'm getting this error on incoming calls on my X100P cards: -- Starting simple switch on 'Zap/1-1' Oct 7 15:49:19 ERROR[74769]: callerid.c:260 callerid_feed: fsk_serie made mylen < 0 (-2) Oct 7 15:49:19 WARNING[74769]: chan_zap.c:5369 ss_thread: CallerID feed failed: Success Oct 7 15:49:19 WARNING[74769]: chan_zap.c:5411 ss_thread: CallerID returned with error on channel

[root@www root]# cat /proc/zaptel/1 Span 1: WCFXO/0 "Wildcard X101P Board 1" 1 WCFXO/0/0 FXSKS (In use) Asterisk CVS-HEAD-02/13/05-00:32:03, Copyright (C) 1999 - 2005 Digium. Feb 15 22:33:48 NOTICE[3002]: callerid.c:307 callerid_feed: Caller*ID failed checksum Feb 15 22:33:51 ERROR[3002]: callerid.c:261 callerid_feed: fsk_serie made mylen < 0 (-6) Feb 15 22:33:51

Hi, I've been looking at the various SIP VoIP service providers and their plans. I understand that Asterisk can be configured as a SIP client to access, for example, a BroadVoice account to access the PSTN and discount LD. I see that a lot of the features provided by SIP VoIP service providers are really not needed since Asterisk will provide them locally. I have no plans on dropping my

Hi, How well to the Cisco 7905G or Cisco 7912G phone work with Asterisk? Cisco claims both phones do SIP. I was strongly considering Polycom phones. However, it appears to be quite difficult to obtain support or firmware for Polycom phones. On the other hand, I find Cisco is very well supported. Thanks, Adi

(hardware in my computer: linux, asterisk, x100p, grandstream budge tone-100 ) Hi, When i run #asterisk ?v It show me a messages but when i try to incomming the call it show me that. Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:3140 sip_reg_timeout: Registration for 'me@192.168.0.6' timed out, trying again Apr 11 07:59:01 NOTICE[81926]: chan_sip.c:5568 handle_request: Registration

Hi, Where can I find information on H.323 for Asterisk and/or integration with Cisco CallManager in particular? <http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration> I have oh323 working on Asterisk. Since the CallManger I am working with is running 3.3.3 I cannot use SIP... Thanks, Adi

I've got a single phone line coming into an X100P. In extensions.conf I've got this: [inboundzap] exten => s,1,Answer exten => s,2,EAgi,hanguptest.agi I see the ring come in and Asterisk detects it and tries to do something with it: NOTICE[20492]: File chan_zap.c, Line 4017 (ss_thread): Got event 2 (Ring/Answered)... -- Executing Answer("Zap/1-1", "") in

I have been experiencing hung up when answering incoming calls through x100p. NOTICE[1242768320]: chan_zap.c:4584 ss_thread: Got event 2 (ring/Answered).. -- Executing Wait("Zap/1-1","1") in new stack -- Executing Answer("Zap/1-1","") in new stack -- Executing DigitTimeout("Zap/1-1"."5") in new stack -- Set digit timeout to 5 --

Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows:

Hi, I've got an x100P and I'm able to dial out and make phone calls with it ok but I just want to set it up to answer the phone and be a simple answering machine but it doesn't seem to want to answer the phone. I keep getting this: on the console when the phone rings: -- Starting simple switch on 'Zap/1-1' Nov 28 08:55:09 NOTICE[29298]: chan_zap.c:5458 ss_thread: Got