Re: [Asterisk-Users] asterisk prompts?
Am 24.05.2004 um 04:36 schrieb hank: hello where can I get the asterisk prompts that are included in the sample config at? they are located in the sounds folder after checkout of the cvs and in /var/lib/asterisk/sounds/ after installing *. Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Prepaid
On Mon, 24 May 2004, usedcanon wrote: I have a requirement for a setup with prepaid call credits. I am aware of the two applications available (been researching for the past week), app_prepaid and app_rateengine. However neither of the two sound like exactly what I want. However I was wondering that someone who has used it might be able to say if they could be used in my scenario. Basically my scenario is pretty straight forward. Credit will be allocated to the ddi, I dont need any announcements etc (maybe low credit warning during call could be useful thoug). From the users prespective everything will be transparent. However the call should disconnect when the credit runs out. The CDR and the account DB need to be adjusted according to the call made. My guess is that app_prepaid could used with modification, I am assuming here that this is not possible as-is with configuration. Basically in case of the prepaid app, the card number can be replace transparently with the callerID. Hi, I did this to app_prepaid - you can pass a parameter into Prepaid() - its looked up in a table to find an associated card number - if that is found then the card number prompt is skipped and the associated card is used automatically. I can send a patch if you like (will also include a minor change or two to have app_prepaid work against CVS. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] STREAM FILE question
Dear all I was wondering is there a way to advance/rewind in playback?(STREAM FILE) say 5 seconds somehow i don't think so but I thought I' would ask Thanks Jer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem receiving a fax with RxFAX
Hey All, I've been trying to get SpanDSP / RxFAX to work in order to set up a soft-fax machine on my asterisk system. I have asterisk CVS-04/08/04-10:06:15 and spandsp 0.0.1. This is on a Fedora Core 1 with 2.4.26 kernel. I have tried to look for a newer version of the spandsp stuff, but opencall.org does not seem to exist in DNS any more. (If it's moved location, can someone please update the Wiki - http://www.voip-info.org/wiki-Asterisk+fax ??) Anyway, I have got it compiled and installed ok. When I try to receive a fax, it does not seem to complete handshaking with the sending machine. The comment from the person sending me the test-fax was that it sounded too fast. Does anyone have any ideas? Calls are coming in via an isdn4linux interface if it would make any difference. Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 2 Sip phones behind un-natted Asterisk
I have 2 SIP phones (Cisco 7960 XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) I have verify=yes in the sip.conf for both phones. Both phones constantly go Unreachable. (However, the connection is very fast between * and sip phones) Sometimes but not always when I try to call phone1 phone2 rings. Is this Nat messing with me or something else? In any caseAny advice out there? Thanks, Chad
[Asterisk-Users] RE: snom reporting busy when it shouldn't
I have no idea what is is, would be great if you can help me. I think this problem is in conjuction with the problem that when i dial the snoms in idle, without answering (exten == s,1,answer) before. Then it is ringing once and then * becoming an busy too. (May be if the 2. ring is coming). Thanks Christian nicolas Christian Stredicke wrote: Did you check if the phone is in DND state? Is there anything strange on the display? CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of nicolas Sent: Sunday, May 23, 2004 5:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] snom reporting busy when it shouldn't I am using asterisk cvs. Incoming/Outgoing calls are working. Calling the phone when some other lines are in use on the phone is ok. What does not work though is when the phone is ringing, nobody else can call the phone anymore. That's what * is saying: -- Got SIP response 486 Busy Here back from 192.168.1.250 -- SIP/snom1-4a44 is busy I am using the 2.05e snom200 firmware. Snom people sad must run. nico ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] updating the dial application
hi Am running asterisk-0.9.0 on a linux slackware box, in which the dial application does not support the S(x) option, am interested in updating this application so it supports this option. Is there anyway to change the application so that the whole system is not affected with regards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk firewall config
If your firewall has some form of sip inspect then you will not need to leave open the rtp ports. Chris - Original Message - From: Tony Hoyle [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 22, 2004 11:11 PM Subject: [Asterisk-Users] Asterisk firewall config The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension numberpassword could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there isn't one documented... Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk firewall config
Ah yes. I too would like to see ip_conntrack_sip :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton Sent: 24 May 2004 08:57 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk firewall config If your firewall has some form of sip inspect then you will not need to leave open the rtp ports. Chris - Original Message - From: Tony Hoyle [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 22, 2004 11:11 PM Subject: [Asterisk-Users] Asterisk firewall config The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension numberpassword could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there isn't one documented... Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
On Mon, 24 May 2004, Chad Brown wrote: 1.The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2.I have verify=yes in the sip.conf for both phones. Both phones constantly go Unreachable. (However, the connection is very fast between * and sip phones) 3.Sometimes but not always when I try to call phone1 phone2 rings. Is this Nat messing with me or something else? In any case...Any advice out there? Yes - I think your NAT firewall is messing with you. I suspect that if you configure the two phones in different ports - IE move one away from 5060, then you'll probably unconfuse your firewall. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Link on Web Pages
On Fri, May 21, 2004 at 01:01:10PM -0400, Barry Fawthrop wrote: In Order to place a call Me button on a webpage which would you use ? [A] a href=sip:[EMAIL PROTECTED]Call Me/a [B] a mailto:sip:[EMAIL PROTECTED]Call Me/a I don't know anything much about SIP/VoIP integration within browsers, but I do know that [A] is valid HTML and [B] is not. So [B] will never work, whereas [A] looks credible. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX generates no tiff file
Good day. I have such problem with rxfax: When I send a fax in fine mode resolution i receive only 50-60% (sometimes 25%) of the page and the rest is compressed lines. http://robik.azhelp.net/1084284449.0.tif http://robik.azhelp.net/1084289786.1.tif Please advise On Mon, 2004-05-24 at 03:03, Steve Underwood wrote: For most people who are sure they have no frame slips, the problem usually turns out to be frame slips :-) If you are *really* sure you do not have frame slips, then uncomment the first line in t30.c, and rebuild and reinstall spandsp. The when you exchange a fax you should end up with a pair of audio files in your /tmp directory - one for the transmit signal and one for the receive signal. Send those to me, and I will investigate. Regards, Steve -- Best regards Vlad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX generates no tiff file
Am 23.05.2004 um 22:58 schrieb Sam Bingner: You should Answer() your calls... In the 5000 exten, you could move your Answer to after the dial if you like... And the h exten hangs up if it doesn't exist so that's redundant, but not bad I have added the Answer() to the extensions but without success. The sender gets the success but I have nothing received. Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Audio Problem
Hi, I have set up the asterisk in Redhat 9 with kphone, kphone is successfully registered under asterisk console.so i had some tests on the extension..but therewere messages saying that something(probably .gsm files)were being played, but i heard nothing.there were also some warning messages from the asterisk console(when i launch it asterisk -vvvc) saying that ERROR:sound device currently not available(some sort like this).may i know what's wrong? 1)do i need to install any codec in order to solve this problem? actually i have downloaded the iLBC source files...but do not how to install it"make install", "./configure"cannot be executableso any ideas? 2)Actually im planning to set the asterisk under cygwin? can this be done?can asterisk be operated under cygwin? thanks alot Do you Yahoo!?Yahoo! Domains - Claim yours for only $14.70/year
Re: [Asterisk-Users] Asterisk firewall config
The latest cisco ios which has ip sip inspect seems to work well. Of course with cisco you swap one set of bugs for another set when you upgrade. I have yet to get a version of the ios that has all the features I want working at the same time:-( Chris - Original Message - From: Karl Dyson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 24, 2004 9:23 AM Subject: RE: [Asterisk-Users] Asterisk firewall config Ah yes. I too would like to see ip_conntrack_sip :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Stenton Sent: 24 May 2004 08:57 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk firewall config If your firewall has some form of sip inspect then you will not need to leave open the rtp ports. Chris - Original Message - From: Tony Hoyle [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, May 22, 2004 11:11 PM Subject: [Asterisk-Users] Asterisk firewall config The asterisk wiki states that it needs SIP, IAX2, IAX and RTP open to the world to work. Is this necessarily true, or does it only need some of these outgoing? I'm concerned as anyone that could guess an extension numberpassword could use my server to make outgoing calls. It would help if the extensions had a netmask/allowable IP setting like the iax.conf file uses, but there isn't one documented... Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk This e-mail has been scanned for all viruses by Star Internet. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem receiving a fax with RxFAX
Hello, I have asterisk CVS-04/08/04-10:06:15 and spandsp 0.0.1. Where did you get it? I'd like to take a chance on it too ;-) I have tried to look for a newer version of the spandsp stuff, but opencall.org does not seem to exist in DNS any more. (If it's moved location, can someone please update the Wiki - http://www.voip-info.org/wiki-Asterisk+fax ??) No, the domain seems not to be moved. The registry has still it's information listed. Is seems that the complete provider located in Hong Kong is disconnected for a longer while so that the TTL is expired. Weird... I could not find any alternatives via google... Regards, Julian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX problems using CVS HEAD, but not CVS STABLE
Hi All, Sorry if this has been covered in the past; I've tried searching the archives, but haven't had any luck finding a similar problem. Basically I have problems when using IAX2 (which I now understand is just IAX). I have three IAX connections setup - VoicePulse, IAXtel, and an Asterisk IAX-PSTN termination provider here in Sydney (ATP) If I try to use the CVS STABLE version of Asterisk, incoming calls from ATP work fine and IAX as a whole works perfectly. However, if I try to use CVS HEAD, incoming calls via ATP don't work (but everything else does). Looking at the debug output produced with iax2 debug, my Asterisk box is prompting the remote ATP Asterisk box for iaxtel authorisation which isn't correct (when using CVS STABLE, it prompts for atp authorisation which is correct). The debug output is below (I have removed phone numbers and hostname): -- begin debug IAX2 Debugging Enabled Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 1ms SCall: 16386 DCall: 0 [snipped:4569] VERSION : 2 CALLED NUMBER : 8231 CALLING NUMBER : 024625 LANGUAGE: en FORMAT : 2 CAPABILITY : 2 ADSICPE : 2 DATE TIME : 146315671 jazz*CLI Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 3ms SCall: 16385 DCall: 16386 [snipped:4569] AUTHMETHODS : 4 CHALLENGE : 207264430 USERNAME: iaxtel jazz*CLI Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 16386 DCall: 16385 [snipped:4569] Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: HANGUP Timestamp: 09160ms SCall: 16386 DCall: 16385 [snipped:4569] Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 09160ms SCall: 16385 DCall: 16386 [snipped:4569] Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 1ms SCall: 00012 DCall: 0 [snipped:4569] Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 1ms SCall: 2 DCall: 00012 [snipped:4569] Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 1ms SCall: 00012 DCall: 2 [snipped:4569] -- end debug Any input would be greatly appreciated. I've included my iax.conf below (based on the sample.conf - many comments have been stripped and usernames removed). Thanks, Shaun -- begin iax.conf [general] port=5036 ;bindaddr=192.168.0.1 bandwidth=high allow=all register = iaxtelusername:iaxtelpassword@iaxtel.com register = vpusername:vppassword@gw.voicepulse.com register = atpusername:atppassword@atpgw tos=lowdelay ; ; Guest sections for unauthenticated connection attempts. Just ; specify an empty secret, or provide no secret section. ; [guest] type=user context=default callerid=Guest IAX User ; ; Trust Caller*ID Coming from iaxtel.com ; [iaxtel] type=user context=incoming-iaxtel auth=rsa inkeys=iaxtel [iaxtel2] ; ; Backwards compatible entry for IAXtel pre-RSA ; type=user context=incoming-iaxtel deny=0.0.0.0/0.0.0.0 permit=69.73.19.178/255.255.255.255 [voicepulse] context = VPWS secret=vpsecret auth=md5 type=friend host=gw.voicepulse.com trunk=no disallow=all allow=gsm [atp] context=atp secret=secret auth=md5 type=friend host=atpgw trunk=yes ;disallow=all ;allow=gsm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SetVar - bellcode and cisco phone
I am trying to have the ring types different for internal and external incoming calls. I have followed the guide on the wiki, the SetVar executes, in extensions.conf I have it as s,1, Yet it doesnt work? When the phone rings, the ring type is the one I chose on the phone, it rings same tone for both when I test. Using Asterisk Stable. Anyone got this working and can point me in the right direction? Ouput of both internal and external incoming calls. -- Executing Macro(SIP/20-5722, stdexten|SIP/22) in new stack -- Executing SetVar(SIP/20-5722, ALERT_INFO=Bellcode-dr2) in new stack -- Executing Dial(SIP/20-5722, SIP/22|25|tr) in new stack -- Called 22 -- SIP/22-080c is ringing == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'SIP/20-5722' in macro 'stdexten' == Spawn extension (sip, 22, 1) exited non-zero on 'SIP/20-5722' -- Executing SetVar(CAPI[contr1/s]/0, ALERT_INFO=Bellcode-dr5) in new stack -- Executing Dial(CAPI[contr1/s]/0, SIP/22|35|t) in new stack -- Called 22 -- started pbx on channel (callgroup=2)! -- SIP/22-e97c is ringing == Spawn extension (incoming, s, 2) exited non-zero on 'CAPI[contr1/s]/0' -- CAPI Hangingup
Re: [Asterisk-Users] PRI problem???
Tim, I would double check the timing. It seems odd that you would supply clock rather than the switch, and if you get clock slips, that could certainly account for what you are seeing. Feel free to contact me off-list if you need more info or have any questions. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Sun, 23 May 2004, Timothy R. McKee wrote: I have just finished installing a new asterisk box at my work. The box is quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4 port Digium T1 card for channel bank and PRI access. I activated a PRI from a local CLEC (DMS-500 based, National protocol). This PRI is on slot 2 of the card and is set as the primary timing source. It is ESF/B8ZS. All the software is latest CVS HEAD as of 5/22/04. My problem lies in random intermittent drops of calls. The entire PRI seems to disappear, dropping all current established calls. I see occasional printouts on an asterisk management console showing all 23 B channels resetting with no reason given: pbx1*CLI -- B-channel 1 successfully restarted on span 2 -- B-channel 2 successfully restarted on span 2 -- B-channel 3 successfully restarted on span 2 -- B-channel 4 successfully restarted on span 2 -- B-channel 5 successfully restarted on span 2 -- B-channel 6 successfully restarted on span 2 -- B-channel 7 successfully restarted on span 2 -- B-channel 8 successfully restarted on span 2 -- B-channel 9 successfully restarted on span 2 -- B-channel 10 successfully restarted on span 2 -- B-channel 11 successfully restarted on span 2 -- B-channel 12 successfully restarted on span 2 -- B-channel 13 successfully restarted on span 2 -- B-channel 14 successfully restarted on span 2 -- B-channel 15 successfully restarted on span 2 -- B-channel 16 successfully restarted on span 2 -- B-channel 17 successfully restarted on span 2 -- B-channel 18 successfully restarted on span 2 -- B-channel 19 successfully restarted on span 2 -- B-channel 20 successfully restarted on span 2 -- B-channel 21 successfully restarted on span 2 -- B-channel 22 successfully restarted on span 2 -- B-channel 23 successfully restarted on span 2 pbx1*CLI (This is from an asterisk console started with -r.) Has anyone encountered similar behavior in the past? If so, what was the resolution? Thanks, Tim McKee configs: zaptel.conf # # span 1 is for channel bank (24 FXS) span=1,2,2,esf,b8zs # span 2 is for NuVox PRI (1-23B, 24D) span=2,1,2,esf,b8zs # span 3 is unused span=3,0,0,esf,b8zs # span 4 is unused span=4,0,0,esf,b8zs # # zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=dialnational ; signalling=fxo_ks use_callerid=yes callwaiting=no restrictcid=no usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes relaxdtmf=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no amaflags=billing channels 1-24 omitted ; ; NuVox PRI context=sdnpri ; switchtype = national pridialplan = unknown signalling = pri_cpe callerid=asreceived group = 2 channel = 25-47 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problem receiving a fax with RxFAX
If you need it I can drop you. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Julian Pawlowski Sent: Monday, May 24, 2004 1:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Problem receiving a fax with RxFAX Hello, I have asterisk CVS-04/08/04-10:06:15 and spandsp 0.0.1. Where did you get it? I'd like to take a chance on it too ;-) I have tried to look for a newer version of the spandsp stuff, but opencall.org does not seem to exist in DNS any more. (If it's moved location, can someone please update the Wiki - http://www.voip-info.org/wiki-Asterisk+fax ??) No, the domain seems not to be moved. The registry has still it's information listed. Is seems that the complete provider located in Hong Kong is disconnected for a longer while so that the TTL is expired. Weird... I could not find any alternatives via google... Regards, Julian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX generates no tiff file
Hi Darren If you are seeing IRQ misses there must be data misses too. :-) Regards, Steve Darren Nickerson wrote: Steve, We have no frame slips, so we probably have a frame slip problem ;-) This may be plaguing us on our faxing. Gain and echo cancelation (ie: none) are all approximately correct, and yet still we cannot get reliable faxing through the POTS lines plugged into our FXO card on the Adit (whereas we can fax well when using the POTS lines directly). Faxing T1 - T1 via a TE405P works well, ... it's only when we try to use the connection to the Adit (24 fxs_ks channels in a T1) that things go horribly wrong. Is there any sure-fire way to detect frame slips? I see a counter for IRQ misses with zttool, but that's all. In my Adit600 I see lots of measures of errors (line errored seconds, controlled slip seconds, bursty errored seconds etc) but they're all zero. Am I missing an obvious way to detect./observe these events? -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 +1.215.243.8335 (fax) - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 8:03 PM Subject: Re: [Asterisk-Users] RxFAX generates no tiff file Hi Petr, For most people who are sure they have no frame slips, the problem usually turns out to be frame slips :-) If you are *really* sure you do not have frame slips, then uncomment the first line in t30.c, and rebuild and reinstall spandsp. The when you exchange a fax you should end up with a pair of audio files in your /tmp directory - one for the transmit signal and one for the receive signal. Send those to me, and I will investigate. Regards, Steve Petr Grussmann wrote: I have same problem connected to PBX over E1 and sync and not slip I have latest version spanDSP I receiving 1/3 pages from faxis ? who is a problems-) I Steve Underwood wrote: Hi Troy, People had a lot of problems like this with earlier versions of spandsp. However, the latest version is pretty solid, and people are using it in high volume production applications. If you are getting these bad results with the latest version I would be interested to see the audio log file, so I can investigate the reason. Regards, Steve Troy Settle wrote: Dunno about not being able to generate a tiff, I got rxfax to do that, but they're badly malformed. http://roanoke-voip01.psknet.com/fax/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI problem???
I thought the timing priority setting was for which incoming timing signal was used as the primary clock source, so I set the PRI as the highest priority clock source. In the telco world this is that way it normally works. Does the priority setting mean something different in zaptel.conf? My apologies to the group re the B channel restarts issue. My searches must have been too specific. Timothy R. McKee -Original Message- From: Bruce Komito [mailto:[EMAIL PROTECTED] Sent: Monday, May 24, 2004 08:30 To: Timothy R. McKee Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PRI problem??? Tim, I would double check the timing. It seems odd that you would supply clock rather than the switch, and if you get clock slips, that could certainly account for what you are seeing. Feel free to contact me off-list if you need more info or have any questions. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Sun, 23 May 2004, Timothy R. McKee wrote: I have just finished installing a new asterisk box at my work. The box is quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4 port Digium T1 card for channel bank and PRI access. I activated a PRI from a local CLEC (DMS-500 based, National protocol). This PRI is on slot 2 of the card and is set as the primary timing source. It is ESF/B8ZS. All the software is latest CVS HEAD as of 5/22/04. My problem lies in random intermittent drops of calls. The entire PRI seems to disappear, dropping all current established calls. I see occasional printouts on an asterisk management console showing all 23 B channels resetting with no reason given: pbx1*CLI -- B-channel 1 successfully restarted on span 2 -- B-channel 2 successfully restarted on span 2 -- B-channel 3 successfully restarted on span 2 -- B-channel 4 successfully restarted on span 2 -- B-channel 5 successfully restarted on span 2 -- B-channel 6 successfully restarted on span 2 -- B-channel 7 successfully restarted on span 2 -- B-channel 8 successfully restarted on span 2 -- B-channel 9 successfully restarted on span 2 -- B-channel 10 successfully restarted on span 2 -- B-channel 11 successfully restarted on span 2 -- B-channel 12 successfully restarted on span 2 -- B-channel 13 successfully restarted on span 2 -- B-channel 14 successfully restarted on span 2 -- B-channel 15 successfully restarted on span 2 -- B-channel 16 successfully restarted on span 2 -- B-channel 17 successfully restarted on span 2 -- B-channel 18 successfully restarted on span 2 -- B-channel 19 successfully restarted on span 2 -- B-channel 20 successfully restarted on span 2 -- B-channel 21 successfully restarted on span 2 -- B-channel 22 successfully restarted on span 2 -- B-channel 23 successfully restarted on span 2 pbx1*CLI (This is from an asterisk console started with -r.) Has anyone encountered similar behavior in the past? If so, what was the resolution? Thanks, Tim McKee configs: zaptel.conf # # span 1 is for channel bank (24 FXS) span=1,2,2,esf,b8zs # span 2 is for NuVox PRI (1-23B, 24D) span=2,1,2,esf,b8zs # span 3 is unused span=3,0,0,esf,b8zs # span 4 is unused span=4,0,0,esf,b8zs # # zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=dialnational ; signalling=fxo_ks use_callerid=yes callwaiting=no restrictcid=no usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes relaxdtmf=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no amaflags=billing channels 1-24 omitted ; ; NuVox PRI context=sdnpri ; switchtype = national pridialplan = unknown signalling = pri_cpe callerid=asreceived group = 2 channel = 25-47 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
I am having exactly the same problem with two phnes connected to a Sipura behind a Linksys. I'm sure this is NAT, because it works fine when I move the Sipura out from behind the Linksys. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, Chad Brown wrote: I have 2 SIP phones (Cisco 7960 XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration... 1.The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2.I have verify=yes in the sip.conf for both phones. Both phones constantly go Unreachable. (However, the connection is very fast between * and sip phones) 3.Sometimes but not always when I try to call phone1 phone2 rings. Is this Nat messing with me or something else? In any case...Any advice out there? Thanks, Chad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is it Possible
Hello I am trying to setup Asterisk on 2 servers PBX300 and PBX200. PBX300 has X100P card with1 telephone line. PBX200 don't have any Zap device. Softphone from PBX200 can talk to softphone on PBX300 but no out going call from PBX200. I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices. I am geting error message " Rejected connect attempt from PBX200". Please help if this is possible. Thanks Deepak
RE: [Asterisk-Users] PRI problem???
My experience has been that the telco is the one that supplies the clock, because they must keep all of their circuits and equipment in sync. In fact, they typically derive their clocking ultimately from a satellite source. At one time, we had a switch interconnected with trunks to the ILEC, and we had to buy a similar clock to drive our equipment so that all of our circuits were in sync with theirs, as well as the other telcos we connected to. Before we did this, our PRIs (which were used for data only) would work for a while, but the modems dialed in to the PRI would eventually drop off, as the clock slippage increased to the point of being out of tolerance. Eventually, the entire PRI had to be restarted. When we installed the clock, all of the problems disappeared. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, Timothy R. McKee wrote: I thought the timing priority setting was for which incoming timing signal was used as the primary clock source, so I set the PRI as the highest priority clock source. In the telco world this is that way it normally works. Does the priority setting mean something different in zaptel.conf? My apologies to the group re the B channel restarts issue. My searches must have been too specific. Timothy R. McKee -Original Message- From: Bruce Komito [mailto:[EMAIL PROTECTED] Sent: Monday, May 24, 2004 08:30 To: Timothy R. McKee Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PRI problem??? Tim, I would double check the timing. It seems odd that you would supply clock rather than the switch, and if you get clock slips, that could certainly account for what you are seeing. Feel free to contact me off-list if you need more info or have any questions. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Sun, 23 May 2004, Timothy R. McKee wrote: I have just finished installing a new asterisk box at my work. The box is quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4 port Digium T1 card for channel bank and PRI access. I activated a PRI from a local CLEC (DMS-500 based, National protocol). This PRI is on slot 2 of the card and is set as the primary timing source. It is ESF/B8ZS. All the software is latest CVS HEAD as of 5/22/04. My problem lies in random intermittent drops of calls. The entire PRI seems to disappear, dropping all current established calls. I see occasional printouts on an asterisk management console showing all 23 B channels resetting with no reason given: pbx1*CLI -- B-channel 1 successfully restarted on span 2 -- B-channel 2 successfully restarted on span 2 -- B-channel 3 successfully restarted on span 2 -- B-channel 4 successfully restarted on span 2 -- B-channel 5 successfully restarted on span 2 -- B-channel 6 successfully restarted on span 2 -- B-channel 7 successfully restarted on span 2 -- B-channel 8 successfully restarted on span 2 -- B-channel 9 successfully restarted on span 2 -- B-channel 10 successfully restarted on span 2 -- B-channel 11 successfully restarted on span 2 -- B-channel 12 successfully restarted on span 2 -- B-channel 13 successfully restarted on span 2 -- B-channel 14 successfully restarted on span 2 -- B-channel 15 successfully restarted on span 2 -- B-channel 16 successfully restarted on span 2 -- B-channel 17 successfully restarted on span 2 -- B-channel 18 successfully restarted on span 2 -- B-channel 19 successfully restarted on span 2 -- B-channel 20 successfully restarted on span 2 -- B-channel 21 successfully restarted on span 2 -- B-channel 22 successfully restarted on span 2 -- B-channel 23 successfully restarted on span 2 pbx1*CLI (This is from an asterisk console started with -r.) Has anyone encountered similar behavior in the past? If so, what was the resolution? Thanks, Tim McKee configs: zaptel.conf # # span 1 is for channel bank (24 FXS) span=1,2,2,esf,b8zs # span 2 is for NuVox PRI (1-23B, 24D) span=2,1,2,esf,b8zs # span 3 is unused span=3,0,0,esf,b8zs # span 4 is unused span=4,0,0,esf,b8zs # # zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=dialnational ; signalling=fxo_ks use_callerid=yes callwaiting=no restrictcid=no usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes relaxdtmf=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no amaflags=billing channels 1-24 omitted ; ; NuVox PRI context=sdnpri ; switchtype = national
[Asterisk-Users] Cisco Asterisk
All, I have access to a Cisco AS5300 w/ 4 T-1's and a Cisco 3600 with no boards. I was wondering if it would be possible some how to Have one of these Ciscos in-between our sip phones and the asterisk server so that we could use G729 Codec. Sip Phones (7960's ATA's) via G729 --Cisco Gateway--Asterisk via G711. Any ideas? Has anybody done such an implementation or know where I could find more information? TIA, Jon
Re: [Asterisk-Users] RxFAX generates no tiff file
Steve, The IRQ miss counter is only 45, and we've sent thousands of faxes since the last boot. I'd like to understand the IRQ miss problem and do what we can to remedy it, but I think that may be separate and distinct from our faxing problems. Faxes fail without the IRQ miss counter incrementing. Anyway, the question I was really looking for you to tackle was 'how do I detect the frame slips you say we all have' ;-) -d -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 +1.215.243.8335 (fax) - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 24, 2004 8:34 AM Subject: Re: [Asterisk-Users] RxFAX generates no tiff file Hi Darren If you are seeing IRQ misses there must be data misses too. :-) Regards, Steve Darren Nickerson wrote: Steve, We have no frame slips, so we probably have a frame slip problem ;-) This may be plaguing us on our faxing. Gain and echo cancelation (ie: none) are all approximately correct, and yet still we cannot get reliable faxing through the POTS lines plugged into our FXO card on the Adit (whereas we can fax well when using the POTS lines directly). Faxing T1 - T1 via a TE405P works well, ... it's only when we try to use the connection to the Adit (24 fxs_ks channels in a T1) that things go horribly wrong. Is there any sure-fire way to detect frame slips? I see a counter for IRQ misses with zttool, but that's all. In my Adit600 I see lots of measures of errors (line errored seconds, controlled slip seconds, bursty errored seconds etc) but they're all zero. Am I missing an obvious way to detect./observe these events? -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 +1.215.243.8335 (fax) - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 8:03 PM Subject: Re: [Asterisk-Users] RxFAX generates no tiff file Hi Petr, For most people who are sure they have no frame slips, the problem usually turns out to be frame slips :-) If you are *really* sure you do not have frame slips, then uncomment the first line in t30.c, and rebuild and reinstall spandsp. The when you exchange a fax you should end up with a pair of audio files in your /tmp directory - one for the transmit signal and one for the receive signal. Send those to me, and I will investigate. Regards, Steve Petr Grussmann wrote: I have same problem connected to PBX over E1 and sync and not slip I have latest version spanDSP I receiving 1/3 pages from faxis ? who is a problems-) I Steve Underwood wrote: Hi Troy, People had a lot of problems like this with earlier versions of spandsp. However, the latest version is pretty solid, and people are using it in high volume production applications. If you are getting these bad results with the latest version I would be interested to see the audio log file, so I can investigate the reason. Regards, Steve Troy Settle wrote: Dunno about not being able to generate a tiff, I got rxfax to do that, but they're badly malformed. http://roanoke-voip01.psknet.com/fax/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI problem???
On Mon, 2004-05-24 at 07:37, Timothy R. McKee wrote: I thought the timing priority setting was for which incoming timing signal was used as the primary clock source, so I set the PRI as the highest priority clock source. In the telco world this is that way it normally works. Does the priority setting mean something different in zaptel.conf? My apologies to the group re the B channel restarts issue. My searches must have been too specific. Tim McKee configs: zaptel.conf # # span 1 is for channel bank (24 FXS) span=1,2,2,esf,b8zs # span 2 is for NuVox PRI (1-23B, 24D) span=2,1,2,esf,b8zs # span 3 is unused span=3,0,0,esf,b8zs # span 4 is unused span=4,0,0,esf,b8zs # # Anything other than a 0 for timing means you are accepting timing from that source. Since you are connected to an outside provider, I would suggest not accepting timing from the channel bank as it doesn't know what timing the outside world is using. So try span=1,0,1,esf,b8zs # don't overdive what is probably a # short hall line. span=2,1,1,esf,b8zs # I think the length here is just # back to the smart jack, -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Variable AGI Parser
is there a way to write uniqueid from call to a varable? Victor Medrano [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with IAX , voice Distortion or Breakage.
Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) System details: - Linux Slackware 9.1 kernel 2.4.6 - CPU PIII 800 - RAM 500 Mb - motherboard Asus TUSL-2c - hard drive IDE - Asterisk last cvs update - 3 TDM400 cards - codec g729 voiceage Help please. May be somebody have same problem. Thank you. -- Best regards, Alexey Ostrovsky Sysadmin Ionidea UA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaprtc-for-2.6
Hello, I just installed * via bri-stuff from junghanns.net. I also use Kernel 2.6.5 and it seems to work fine. I saw the directory zaprtc-for-2.6 coming with bri-stuff and noticed that it is not used by the install scripts. I have absolutely no idea what this software does. Can anybody clear me up? Regards, Julian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
I have 2 SIP phones (Cisco 7960 XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration... 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both phones. Both phones constantly go Unreachable. (However, the connection is very fast between * and sip phones) 3. Sometimes but not always when I try to call phone1 phone2 rings. Have you tried to make sure that each user agent use differnet sip port? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with IAX , voice Distortion or Breakage.
On Mon, 2004-05-24 at 08:27, Alexey Ostrovsky wrote: Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) System details: - Linux Slackware 9.1 kernel 2.4.6 - CPU PIII 800 - RAM 500 Mb A sysadmin that is so inaccurate as to say 500Mb instead of the obvious 512 Mb? - motherboard Asus TUSL-2c - hard drive IDE - Asterisk last cvs update - 3 TDM400 cards - codec g729 voiceage Help please. May be somebody have same problem. Can you verify you are using g729. Also what kind of network are you doing the IAX call over? What else is on the network at the time of the call. Most of what you say sounds a lot like you are running short of bandwidth. Sounds like the classic problem of running out of the buffered content and decoding what you have when you get it. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
I'm not the original poster, but I have the same problem with a Sipura. In my configuration, I have line 1 set to port 5060 and line 2 set to port 5061. I assume that is what you are suggesting, right? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, Senad Jordanovic wrote: I have 2 SIP phones (Cisco 7960 XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration... 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both phones. Both phones constantly go Unreachable. (However, the connection is very fast between * and sip phones) 3. Sometimes but not always when I try to call phone1 phone2 rings. Have you tried to make sure that each user agent use differnet sip port? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX generates no tiff file
Hi Darren, There is no guarantee your problem is frame slips. However, a number of people have reported problems with rxfax going wrong in the middle of a page. A number have reported other fax hardware doing the same thing. Every one I have investigated so far has been the same problem - frame slips. It is a puzzle to me how these had previously gone unnoticed, since they usually produce an annoying tick each time a slip occurs. However, it seems a lot of people are operating their trunks like that. How to fix them? This is left as an exercise for the reader.. Seriously, there are too many differences between people's setups for me to really say. If you have a channel bank, * should be the clock master for that channel bank. Try checking that it is. Regards, Steve Darren Nickerson wrote: Steve, The IRQ miss counter is only 45, and we've sent thousands of faxes since the last boot. I'd like to understand the IRQ miss problem and do what we can to remedy it, but I think that may be separate and distinct from our faxing problems. Faxes fail without the IRQ miss counter incrementing. Anyway, the question I was really looking for you to tackle was 'how do I detect the frame slips you say we all have' ;-) -d -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 +1.215.243.8335 (fax) - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 24, 2004 8:34 AM Subject: Re: [Asterisk-Users] RxFAX generates no tiff file Hi Darren If you are seeing IRQ misses there must be data misses too. :-) Regards, Steve Darren Nickerson wrote: Steve, We have no frame slips, so we probably have a frame slip problem ;-) This may be plaguing us on our faxing. Gain and echo cancelation (ie: none) are all approximately correct, and yet still we cannot get reliable faxing through the POTS lines plugged into our FXO card on the Adit (whereas we can fax well when using the POTS lines directly). Faxing T1 - T1 via a TE405P works well, ... it's only when we try to use the connection to the Adit (24 fxs_ks channels in a T1) that things go horribly wrong. Is there any sure-fire way to detect frame slips? I see a counter for IRQ misses with zttool, but that's all. In my Adit600 I see lots of measures of errors (line errored seconds, controlled slip seconds, bursty errored seconds etc) but they're all zero. Am I missing an obvious way to detect./observe these events? -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 +1.215.243.8335 (fax) - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 8:03 PM Subject: Re: [Asterisk-Users] RxFAX generates no tiff file Hi Petr, For most people who are sure they have no frame slips, the problem usually turns out to be frame slips :-) If you are *really* sure you do not have frame slips, then uncomment the first line in t30.c, and rebuild and reinstall spandsp. The when you exchange a fax you should end up with a pair of audio files in your /tmp directory - one for the transmit signal and one for the receive signal. Send those to me, and I will investigate. Regards, Steve Petr Grussmann wrote: I have same problem connected to PBX over E1 and sync and not slip I have latest version spanDSP I receiving 1/3 pages from faxis ? who is a problems-) I Steve Underwood wrote: Hi Troy, People had a lot of problems like this with earlier versions of spandsp. However, the latest version is pretty solid, and people are using it in high volume production applications. If you are getting these bad results with the latest version I would be interested to see the audio log file, so I can investigate the reason. Regards, Steve Troy Settle wrote: Dunno about not being able to generate a tiff, I got rxfax to do that, but they're badly malformed. http://roanoke-voip01.psknet.com/fax/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Aastra ADSI phone
I've received my Aastra 390 phone. I got the unlock procedure from the vendor, and the services button now shows all four entrys as available. When I give the phone ADSIProg() the phone displays: Asterisk PBX download refused Conflict with: available The asterisk.adsi hasn't been changed: DESCRIPTION Asterisk PBX VERSION 0x02 SECURITY 0x FDN 0x000f ... When I change the FDN number to anything else, the phone replies Services full. Is the phone properly unlocked? Has anyone seen this before? Yes. The security and FDN codes are incorrect (at least for your phone) in the script. I just added a patch to bugs to fix this. Mark also just updated the VM code to include the proper security and FDN code for the Comedian Mail script (went into CVS Sun 5/23/2004 8:31 PM) so you may want to update your VM as well. Patch your ADSI script with the patch here: http://bugs.digium.com/bug_view_page.php?bug_id=0001709 Then re-run the ADSIProg and it should work. NOTE THAT THIS APPLIES _ONLY_ TO THE Sayson Aastra 380 and 480 PHONES. OTHER ADSI PHONES MAY BE HARDWARE LOCKED AND/OR MAY USE DIFFERENT FDN VALUES. Good luck, Steven Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Asterisk
I have the setup a little different: Phones Asterisk === Gw Works just fine, All you have to do is setup a good dial-plan in asterisk to fwd the calls to you gw. On Mon, 2004-05-24 at 08:53, Jon wrote: All, I have access to a Cisco AS5300 w/ 4 T-1's and a Cisco 3600 with no boards. I was wondering if it would be possible some how to Have one of these Ciscos in-between our sip phones and the asterisk server so that we could use G729 Codec. Sip Phones (7960's ATA's) via G729 --Cisco Gateway--Asterisk via G711. Any ideas? Has anybody done such an implementation or know where I could find more information? TIA, Jon -- Pablo Endres [EMAIL PROTECTED] ComVoz Comunications ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] detecting pick up?
hello is there any way so that i can detect if the called party picked up? with regards tareq ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_capi 0.3.1 , Asterisk , 3 x C4 active ISDN card Segmentation fault
Hi, i use chan_capi 0.3.1 with asterisk (stable branch cvs) and 3 x c4 active ISDN card. From Controller 1 - 7 there are no problems making calls between asterisk and the pstn. But when i make calls from controller 8 - 12 i get on every controller (8 - 12) a segmentation fault in asterisk :( I tried different linux distributions (gentoo 2004.1, redhat 9.0 , suse 9.1) but same error. Example : I call the number on the 8 controller (9766) - there i have a playback - serverproblem.gsm when i hang up i get the segmentation fault: /usr/sbin/asterisk -vvdc with gdb CLI capi info Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. Contr3: 2 B channels total, 2 B channels free. Contr4: 2 B channels total, 2 B channels free. Contr5: 2 B channels total, 2 B channels free. Contr6: 2 B channels total, 2 B channels free. Contr7: 2 B channels total, 2 B channels free. Contr8: 2 B channels total, 2 B channels free. Contr9: 2 B channels total, 2 B channels free. Contr10: 2 B channels total, 2 B channels free. Contr11: 2 B channels total, 2 B channels free. Contr12: 2 B channels total, 2 B channels free. *CLI May 24 07:48:13 DEBUG[1109818288]: channel.c:1493 ast_set_write_format: Set channel CAPI[contr8/97166]/0 to write format ALAW May 24 07:48:13 DEBUG[1109818288]: pbx.c:1739 ast_pbx_run: Spawn extension (default,i,1) exited non-zero on 'CAPI[contr8/97166]/0' May 24 07:48:13 DEBUG[1109818288]: channel.c:662 ast_hangup: Hanging up channel 'CAPI[contr8/97166]/0' sent INFO_RESP (PLCI=0x108) == DISCONNECT_B3_IND NCCI=0x10108 Urgent handler sent DISCONNECT_B3_RESP NCCI=0x10108 -- CAPI Hangingup activehangingup sent DISCONNECT_REQ PLCI=0x108 Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 1084259248 (LWP 3118)] 0x407e81f1 in pipe_msg (PLCI=Variable PLCI is not available.) at chan_capi.c:1319 1319 capi_controllers[p-i-controller]-nfreebchannels++; (gdb) bt #0 0x407e81f1 in pipe_msg (PLCI=Variable PLCI is not available.) at chan_capi.c:1319 #1 0x407e98e4 in do_monitor (data=0x0) at chan_capi.c:2182 #2 0x4002a9dd in start_thread () from /lib/tls/libpthread.so.0 #3 0x40166ffa in clone () from /lib/tls/libc.so.6 (gdb) Is chan_capi limited to 1 x C4 ? Thanks for your help. Kind regards Sascha ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX2 NAT / Registration Issue
Subject: RE: [Asterisk-Users] IAX2 NAT / Registration Issue His firewall is stateless. I've run into the same issue w/the sonic wall firewall on a client site. TL Todd, Did you find a way to alter the configuration of the firewall/NAT to enable state maintenance? The system appears to maintain a least some kind of state, as the IAX Phone can receive calls for 15 - 30 seconds after completing an outbound call. Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFAX generates no tiff file
Steve, Thanks, that's helpful. Please let me be clear, I'm not asking for a way to fix them, but rather a way to see if slips are happening in the first place. The annoying tick is helpful information ... we can listen for that. I was hoping a more diagnostic way might exist. I guess not! ;-) -d - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 24, 2004 9:55 AM Subject: Re: [Asterisk-Users] RxFAX generates no tiff file Hi Darren, There is no guarantee your problem is frame slips. However, a number of people have reported problems with rxfax going wrong in the middle of a page. A number have reported other fax hardware doing the same thing. Every one I have investigated so far has been the same problem - frame slips. It is a puzzle to me how these had previously gone unnoticed, since they usually produce an annoying tick each time a slip occurs. However, it seems a lot of people are operating their trunks like that. How to fix them? This is left as an exercise for the reader.. Seriously, there are too many differences between people's setups for me to really say. If you have a channel bank, * should be the clock master for that channel bank. Try checking that it is. Regards, Steve Darren Nickerson wrote: Steve, The IRQ miss counter is only 45, and we've sent thousands of faxes since the last boot. I'd like to understand the IRQ miss problem and do what we can to remedy it, but I think that may be separate and distinct from our faxing problems. Faxes fail without the IRQ miss counter incrementing. Anyway, the question I was really looking for you to tackle was 'how do I detect the frame slips you say we all have' ;-) -d -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 +1.215.243.8335 (fax) - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 24, 2004 8:34 AM Subject: Re: [Asterisk-Users] RxFAX generates no tiff file Hi Darren If you are seeing IRQ misses there must be data misses too. :-) Regards, Steve Darren Nickerson wrote: Steve, We have no frame slips, so we probably have a frame slip problem ;-) This may be plaguing us on our faxing. Gain and echo cancelation (ie: none) are all approximately correct, and yet still we cannot get reliable faxing through the POTS lines plugged into our FXO card on the Adit (whereas we can fax well when using the POTS lines directly). Faxing T1 - T1 via a TE405P works well, ... it's only when we try to use the connection to the Adit (24 fxs_ks channels in a T1) that things go horribly wrong. Is there any sure-fire way to detect frame slips? I see a counter for IRQ misses with zttool, but that's all. In my Adit600 I see lots of measures of errors (line errored seconds, controlled slip seconds, bursty errored seconds etc) but they're all zero. Am I missing an obvious way to detect./observe these events? -Darren -- Darren Nickerson Senior Sales Support Engineer iFax Solutions, Inc. www.ifax.com [EMAIL PROTECTED] +1.215.438.4638 +1.215.243.8335 (fax) - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 8:03 PM Subject: Re: [Asterisk-Users] RxFAX generates no tiff file Hi Petr, For most people who are sure they have no frame slips, the problem usually turns out to be frame slips :-) If you are *really* sure you do not have frame slips, then uncomment the first line in t30.c, and rebuild and reinstall spandsp. The when you exchange a fax you should end up with a pair of audio files in your /tmp directory - one for the transmit signal and one for the receive signal. Send those to me, and I will investigate. Regards, Steve Petr Grussmann wrote: I have same problem connected to PBX over E1 and sync and not slip I have latest version spanDSP I receiving 1/3 pages from faxis ? who is a problems-) I Steve Underwood wrote: Hi Troy, People had a lot of problems like this with earlier versions of spandsp. However, the latest version is pretty solid, and people are using it in high volume production applications. If you are getting these bad results with the latest version I would be interested to see the audio log file, so I can investigate the reason. Regards, Steve Troy Settle wrote: Dunno about not being able to generate a tiff, I got rxfax to do that, but they're badly malformed. http://roanoke-voip01.psknet.com/fax/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] Help with IAX , voice Distortion or Breakage.
Steven Critchfield wrote: On Mon, 2004-05-24 at 08:27, Alexey Ostrovsky wrote: Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) System details: - Linux Slackware 9.1 kernel 2.4.6 - CPU PIII 800 - RAM 500 Mb A sysadmin that is so inaccurate as to say 500Mb instead of the obvious 512 Mb? I think there is no differents for my problem. Thank you a lot. - motherboard Asus TUSL-2c - hard drive IDE - Asterisk last cvs update - 3 TDM400 cards - codec g729 voiceage Help please. May be somebody have same problem. Can you verify you are using g729. Also what kind of network are you doing the IAX call over? What else is on the network at the time of the call. Most of what you say sounds a lot like you are running short of bandwidth. Sounds like the classic problem of running out of the buffered content and decoding what you have when you get it. Yes of course I am using g729b codec. I have SHDSL line with 128 bit/c bandwidth. Nothing else on the network in testing time. -- Best regards, Alexey Ostrovsky Sysadmin Ionidea UA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with IAX , voice Distortion or Breakage.
We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) Without more technical data, its hard to guess. Might consider... 1. if you're using cisco phones, upgrade * to current Head or Stable on both ends of the iax link. The Stable code was just fixed on Friday. (There could be other phones/adapters that are impacted by the iax/gsm timestamp problems.) I've also noticed a fair number of other fixes that were just recently applied to Head. 2. if not cisco phones, check the config's on whatever phone you're using to ensure transmit silence is enabled. If you are using the xten soft phone, the parameter is in the Advanced Settings area. 3. Check to ensure all ethernet nic adapters (and associated switch ports) are running in full-duplex mode, etc. 4. If none of the above apply, it would be helpful to see a packet trace (using ethereal) at about the time the distorting/failure is occurring. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
Chad Brown wrote: I have 2 SIP phones (Cisco 7960 XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both phones. Both phones constantly go Unreachable. (However, the connection is very fast between * and sip phones) 3. Sometimes but not always when I try to call phone1 phone2 rings. Is this Nat messing with me or something else? In any caseAny advice out there? Thanks, Chad The problem is probably that both of your SIP phones are using the same port. I played with two 7960's behind a Linksys on Saturday and finally got them playing right when I changed the following: In Phone 1's SIP[macaddr].cnf: voip_control_port: 5061 In Phone 2's SIP[macaddr].cnf: voip_control_port: 5062 The default control port is 5060. Note: This is the port that the PHONE uses to initiate the connection to * and not the port it is connecting to. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapata.conf setup for TE410P
On Sun, 23 May 2004, William Zhang waxed: Hi, I have a TE410P with 3 E1 being enabled, some how it crashes for 2 times lately, I suspect it might be the channel setup issue, can Does it crash immediately or after a fixed amount of time ? anyone tell me if following part in zapata.conf is correct? switchtype = euroisdn signalling = pri_cpe pridialplan=local group = 1 context = incoming channel = 1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124 These channels look fine for E1. You said 3, but there are 4 configured right there. Shouldn't matter much. What does your zaptel.conf look like ? Can you post that ? Also, is there way to log the reason why Asterisk is crashed? Thank you. asterisk -vvgc That will give you a core file and lots of output, plus connect you to asterisk so you can watch it running on a console. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZapRas problems
Hi I try to use zapras. I am using zaptel-bri-0.0.2 I compiled and patched pppd from ftp://ftp.digium.com/pub/zaptel/misc/ pppd is /usr/sbin/pppd Any idea whats going wrong here? Thomas -- Accepting call from '95' to '8526' on channel 1, span 1 -- AGI Script nuller.agi completed, returning 0 -- Executing ZapRAS(Zap/1-1, debug|64000|noauth|netmask|255.255.255.0|192.168.1.121:192.168.1.122) in new stack -- Starting RAS on Zap/1-1 May 24 16:21:00 WARNING[561180]: app_zapras.c:143 run_ras: wait4 returned -1: No child processes -- RAS on Zap/1-1 terminated with signal 1 == Spawn extension (incoming, 8526, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with IAX , voice Distortion or Breakage.
On Mon, 2004-05-24 at 09:39, Alexey Ostrovsky wrote: Steven Critchfield wrote: On Mon, 2004-05-24 at 08:27, Alexey Ostrovsky wrote: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) - Asterisk last cvs update - 3 TDM400 cards - codec g729 voiceage Can you verify you are using g729. Also what kind of network are you doing the IAX call over? What else is on the network at the time of the call. Most of what you say sounds a lot like you are running short of bandwidth. Sounds like the classic problem of running out of the buffered content and decoding what you have when you get it. Yes of course I am using g729b codec. I have SHDSL line with 128 bit/c bandwidth. Nothing else on the network in testing time. I guess I should have been more specific in asking you to provide a actual call log so we can see for sure that asterisk choose to use the G729 codec. It also may shed light on the problem in case it is something else. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] detecting pick up?
On Mon, 2004-05-24 at 09:07, tareq wrote: hello is there any way so that i can detect if the called party picked up? What type of interface? If analog, you should have found it in the archives as a no. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco Asterisk
Actually the reason I need it setup this way, is so that we can use g729 codec. Thats why I want the phone to go thru the cisco first. The cisco will basically just be coding decoding g729 and passing the data to asterisk. The problem I'm having is with authentication. I want the authentication info to be passed thru cisco to asterisk so that asterisk can authenticate w/ sip.conf Is this possible? - Original Message - From: Pablo Endres [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 24, 2004 10:07 AM Subject: Re: [Asterisk-Users] Cisco Asterisk I have the setup a little different: Phones Asterisk === Gw Works just fine, All you have to do is setup a good dial-plan in asterisk to fwd the calls to you gw. On Mon, 2004-05-24 at 08:53, Jon wrote: All, I have access to a Cisco AS5300 w/ 4 T-1's and a Cisco 3600 with no boards. I was wondering if it would be possible some how to Have one of these Ciscos in-between our sip phones and the asterisk server so that we could use G729 Codec. Sip Phones (7960's ATA's) via G729 --Cisco Gateway--Asterisk via G711. Any ideas? Has anybody done such an implementation or know where I could find more information? TIA, Jon -- Pablo Endres [EMAIL PROTECTED] ComVoz Comunications ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
John, In my case, the two ports are not using the same IP port (one is on 5060, the other on 5061), but of course, they are on the same IP address. I think that is what is confusing the NAT server, but I don't know what to do about it. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, John Fraizer wrote: Chad Brown wrote: I have 2 SIP phones (Cisco 7960 XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both phones. Both phones constantly go Unreachable. (However, the connection is very fast between * and sip phones) 3. Sometimes but not always when I try to call phone1 phone2 rings. Is this Nat messing with me or something else? In any case Any advice out there? Thanks, Chad The problem is probably that both of your SIP phones are using the same port. I played with two 7960's behind a Linksys on Saturday and finally got them playing right when I changed the following: In Phone 1's SIP[macaddr].cnf: voip_control_port: 5061 In Phone 2's SIP[macaddr].cnf: voip_control_port: 5062 The default control port is 5060. Note: This is the port that the PHONE uses to initiate the connection to * and not the port it is connecting to. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: setting the number of rings befor asterisk picks up?
- - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. - Original Message - From: hank [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 6:19 PM Subject: setting the number of rings befor asterisk picks up? hello how do I set the number of rings picks up on? I am using a single port fxo card and currently asterisk is answering after 1 or 2 rings and I want it answering after 4 5 or 6 rings thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: creating a single user voice mail box on asterisk?
- - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. - Original Message - From: hank [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 3:56 PM Subject: creating a single user voice mail box on asterisk? hello how do I go create a single boice mail box on asterisk? thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme Options (new one)
Is it possible to select the audio stream thats played as a user enters a meetme conference? If you could, it would be very simple to record a users name, and then play that as the greeting to other attendees as they join the conference. If not, could someone tell me how hard it would be to modify the source? I presume at the moment the file to be played it stored in a var somewhere, is it simply a case of allowing MeetMe() to accept another param, which could be the audio stream? Cheers, Ben Merrills
[Asterisk-Users] zapata ? question
Hello, I want to install t100p. Do I need to install zapata ? I know that I need to install zaptel but what I am not sure if I need zapata. This card is going to be connected to channels bank. Could somebody tell me what is the difference between zapata and zaptel? B ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote Disconnects
Using a Wildcard TE410P, I have span2 setup. Consistently an incoming call that is dropped by the remote party, * can not detect the dropped call and holds it till finally it times out and drops it that way. Here is the config for it: span=2,1,0,d4,ami fxsks=25-48 group=3 callerid=TEst 555-12125 signalling=fxs_ks context=incoming_t1 channel=25-48 usecallerid=no busydetect=yes busycount=10 callprogress=no -- respectfully, Joseph - --= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with IAX , voice Distortion or Breakage
Title: Re: [Asterisk-Users] Help with IAX , voice Distortion or Breakage I am having almost the exact same problem. I have the following setup: Debian Woody Kernel 2.4.18 CPU: P4 1.2GHz RAM: 1GB Asterisk Latest CVS 1 TDM400P card Codec GSM Ive been chasing down bandwidth issues, but have had no luck. We are still pursuing those issues. I just started configuring IAX, so I assumed it was related to my IAX configs. We just noticed this morning that SIP is having the same issue, but that it isnt as severe. We are sending our calls over the regular Internet, but that hasnt been a problem in the past. Besides the problem is too regular to be Internet related (unless it is at my service provider). Regards, John
Re: [Asterisk-Users] Meetme Options (new one)
On May 24, 2004, at 8:21 AM, Ben Merrills wrote: x-tad-biggerIs it possible to select the audio stream thats played as a user enters a meetme conference?/x-tad-bigger I was just now doing an RTFS trying to figure that out. At the moment, the sound played on entering is hard-coded. Time for a feature request?
RE: [Asterisk-Users] Fw: creating a single user voice mail box on asterisk?
You'll probably be ignored on this list -- you fired off three basic questions in an hour which showed that you haven't taken the time to attempt answering those questions yourself. Additionally, you forwarded those same questions without modification or further info. Asterisk is FREE and thus requires some of your time and expertise to set it up. Search google for the information you need, it's readily available. Check www.voip-info.org, www.asterisk.org, www.asteriskdocs.org for more info. You won't be judged because you are blind, but because you chose not to look. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Monday, May 24, 2004 10:17 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fw: creating a single user voice mail box on asterisk? - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. - Original Message - From: hank [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 3:56 PM Subject: creating a single user voice mail box on asterisk? hello how do I go create a single boice mail box on asterisk? thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting indicator do not work for me.
At 02:36 PM 5/22/2004 +0200, nicolas wrote: Hi, The call waiting indicator do not work for me. I am using a snom200 cwi is switched on in phone-config. Have asked snom, but there are can not help me, because it is working for them. When it is coming in an call while the phone is busy. The phone returns: -- Got SIP response 486 Busy Here back from 190.100.200.19 But it should not, should make a call waiting indication. (The same behaviour is when i am dialing the phone (in idle) from extern without making an exten = s,x,Answer.) Hi Nicolas, We experienced the same problem recently with our snom200. It happened when we were trying to upgrade to the 2.05x firmware releases. I believe what happened during one of the many restarts and reloads, a phone option got reset. Try opening up the browser interface to the phone, then clicking SettingsRedirection, then using the Event menu to set to Never. Ours somehow got set to Always. Once we made this change, the busy messages went away. I don't know what the default value for this setting is. As for MWI, it has been our experience that it does not work properly in the 2.04x firmware (MWI light lights but never gets cleared when all messages have been deleted.) We tried updating to all the 2.05x releases which did fix the MWI behavior but broke the flash/hold feature. We went back to 2.04 because hold was more important than a working MWI. YMMV. Michael Swan Neon Software, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip call using name in sip.conf
Hi!
Now, going the other way around is more difficult. #1 doesn't know
the IP address of #2. There is the concept of register= in
sip.conf, but that only registers _individual user-agents_ and does
not allow one server to know that another server is at a particular
IP address.
Hm, are you sure, or did I just misunderstand the problem? In the past I
had some trouble with the reliability of IAX registration, so I introdued
SIP registration for server B as a backup, and it works just fine:
* server B (dynamic):
register = ServBsip:[EMAIL PROTECTED]
* server A (static):
exten = _0XXX,2,Dial(SIP/ServBsip/${EXTEN:1},30,r)
[ServBsip]
context=from-ServBsip
type=friend
username=ServBsip
fromuser=ServBsip
secret=pw
auth=md5
host=dynamic
nat=no
canreinvite=yes
disallow=all
allow=gsm
Feel free to split type=friend into type=peer and type=user and to
introduce deny/permit entries for the hostname/ip.
Cheers, Philipp
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RE: [Asterisk-Users] PRI problem???
Hi Tim- Except for maybe eliminating the channel bank as a secondary source of timing, your conf files look fine. I notice that you've used 2 as the build-out (line length) parameter - I take it that your asterisk box is not right next to the switch. Anyway, I've never seen build-out make much difference, but maybe that's an issue. Although asterisk restarts B channels on a periodic basis, it's not supposed to do this if there's a call in progress, so there's something wrong. I noticed a problem recently that was causing D channels to drop under load, but after reporting this, Mark S fixed this on May 18th, so your more recent distr should cover it. Finally, in looking through libpri bugs on the bug list, Mark refers to some kind of firmware problem on some TE410P's (I think), so you might read through recent bug reports and see if any match your symptom (read the closed ones too) Is there anything remarkable in your /var/log/asterisk/messages log file? regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Timothy R. McKee Sent: Sunday, May 23, 2004 7:52 PM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] PRI problem??? I have just finished installing a new asterisk box at my work. The box is quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4 port Digium T1 card for channel bank and PRI access. I activated a PRI from a local CLEC (DMS-500 based, National protocol). This PRI is on slot 2 of the card and is set as the primary timing source. It is ESF/B8ZS. All the software is latest CVS HEAD as of 5/22/04. My problem lies in random intermittent drops of calls. The entire PRI seems to disappear, dropping all current established calls. I see occasional printouts on an asterisk management console showing all 23 B channels resetting with no reason given: configs: zaptel.conf # # span 1 is for channel bank (24 FXS) span=1,2,2,esf,b8zs # span 2 is for NuVox PRI (1-23B, 24D) span=2,1,2,esf,b8zs # span 3 is unused span=3,0,0,esf,b8zs # span 4 is unused span=4,0,0,esf,b8zs # # zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=dialnational ; signalling=fxo_ks use_callerid=yes callwaiting=no restrictcid=no usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes relaxdtmf=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no amaflags=billing channels 1-24 omitted ; ; NuVox PRI context=sdnpri ; switchtype = national pridialplan = unknown signalling = pri_cpe callerid=asreceived group = 2 channel = 25-47 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
What does the Xten diagnostic log say about a single session? Also, what does the * SIP debug output say? I'd be very interested to see what IPs and ports SIP is trying to set the RTP connection on. (Since SIP appears to be working fine, it's the RTP part that is breaking). Are both the Xten and the 7960 trying to listen on the same RTP port (my Xten is configured to listen on 8000)? Pardon me if I sound like an idiot, but I'm somewhat new to VoIP, SIP _and_ Asterisk. :) Shaun --- Bruce Komito [EMAIL PROTECTED] wrote: John, In my case, the two ports are not using the same IP port (one is on 5060, the other on 5061), but of course, they are on the same IP address. I think that is what is confusing the NAT server, but I don't know what to do about it. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, John Fraizer wrote: Chad Brown wrote: I have 2 SIP phones (Cisco 7960 XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both phones. Both phones constantly go Unreachable. (However, the connection is very fast between * and sip phones) 3. Sometimes but not always when I try to call phone1 phone2 rings. Is this Nat messing with me or something else? In any case Any advice out there? Thanks, Chad The problem is probably that both of your SIP phones are using the same port. I played with two 7960's behind a Linksys on Saturday and finally got them playing right when I changed the following: In Phone 1's SIP[macaddr].cnf: voip_control_port: 5061 In Phone 2's SIP[macaddr].cnf: voip_control_port: 5062 The default control port is 5060. Note: This is the port that the PHONE uses to initiate the connection to * and not the port it is connecting to. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Domains Claim yours for only $14.70/year http://smallbusiness.promotions.yahoo.com/offer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: creating a single user voice mail box on asterisk?
I wasn't sure if my messages were getting threw my apologese thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 24, 2004 9:25 AM Subject: RE: [Asterisk-Users] Fw: creating a single user voice mail box on asterisk? You'll probably be ignored on this list -- you fired off three basic questions in an hour which showed that you haven't taken the time to attempt answering those questions yourself. Additionally, you forwarded those same questions without modification or further info. Asterisk is FREE and thus requires some of your time and expertise to set it up. Search google for the information you need, it's readily available. Check www.voip-info.org, www.asterisk.org, www.asteriskdocs.org for more info. You won't be judged because you are blind, but because you chose not to look. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of hank Sent: Monday, May 24, 2004 10:17 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Fw: creating a single user voice mail box on asterisk? - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. - Original Message - From: hank [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 3:56 PM Subject: creating a single user voice mail box on asterisk? hello how do I go create a single boice mail box on asterisk? thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP local loop?
Are you guys aware of any providers that do IP local loop service? What I want is to get a T-1 from said provider, plug it into my Cisco router, speak SIP to a voice gateway upstream, and have phone calls go out over PSTN from there. This is kind of what Vonage and ATT CallVantage do, but they are more geared toward the residential market, and I want to be able to bring an arbritary number of lines in. thanks, Shaun __ Do you Yahoo!? Yahoo! Domains Claim yours for only $14.70/year http://smallbusiness.promotions.yahoo.com/offer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and OH323
[EMAIL PROTECTED] wrote:
Hello,
i want to use asterisk as a gateway for H323-Phones.
But i cant get it work.
I'm using a gatekeeper on another computer. My IP-phone is registered there.
Does anybody can sent me an oh323.conf and extension.conf as examples?
Thanks in advance
Erik Bastian
In [general] section of oh323.conf enter the IP address of your
gatekeeper:
gatekeeper=gatekeeper-ip
context=default-h323
...leave default values for other keys.
In [register] section of oh323.conf enter the aliases/numbers/prefixes
that are handled by your Asterisk, e.g:
alias=111
alias=112
gwprefix=7
grprefix=6
In [codecs] section configure the encoding to be used by the OH323
channels (XXX define only one codec!), e.g:
codec=G711A
frames=20
Now, in your extensions.conf create the extensions entries which will
handle your incoming calls (those defined in the [register] section), e.g:
[default-h323] ;The context was defined in oh323.conf,
;[general] section
exten = 111,1,Playback(...)
exten = 112,1,Dial(SIP/snom1)
exten = _6XX,1,Dial(OH323/${EXTEN:1},20,tT)
Now you are ready to call your Asterisk by your H.323 IP phones
by just dialing one of the numbers you defined in the [register]
section (e.g. valid entries are: 111,112,700,78910,666,...).
Hope that helps,
Michael.
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Re: [Asterisk-Users] Asterisk-oh323 0.6.1 Compiling problem
I need the full output for this (the first lines are missing). Michael. Nicholas Ruddick wrote: ok done, but now i'm getting different errors - /usr/src/pwlib/include/ptlib/args.h:389: virtual outside class declaration /usr/src/pwlib/include/ptlib/args.h:389: non-member function `UnknownOption (...)' cannot have `const' method qualifier [snip...] in this scope /usr/src/pwlib/include/ptlib/indchan.h:259: `readChannel' was not declared in this scope /usr/src/pwlib/include/ptlib/indchan.h:261: `PChannel' was not declared in this scope /usr/src/pwlib/include/ptlib/indchan.h:261: `writeChannel' was not declared in this scope /usr/src/pwlib/include/ptlib/indchan.h:263: parse error before `=' /usr/src/pwlib/include/ptlib/indchan.h:265: `BOOL Open (...)' redeclared as different kind of symbol /usr/src/pwlib/include/ptlib/indchan.h:229: previous declaration of `BOOL Open' /usr/src/pwlib/include/ptlib/indchan.h:229: previous non-function declaration `BOOL Open' /usr/src/pwlib/include/ptlib/indchan.h:265: conflicts with function declaration `BOOL Open (...)' /usr/src/pwlib/include/ptlib/indchan.h:265: confused by earlier errors, bailing out make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.1/wrapper' make: *** [subdirs_all] Error 1 Whats this all about, it's still complaining about some audio thing i just can't work out. I'm using redhat 7.3 btw, i have both the openh323, pwlib standard, devel and src packages install. Still no joy. Thanks, Nicholas Ruddick Pablo Endres wrote: Check your README file again. In order to compile 0.6.1 you need newer versions of pwlib and openh323 (1.6.6 and 1.13.5) Then it should work just fine Pablo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP local loop?
On Mon, 2004-05-24 at 12:19, Shaun Dawson wrote: Are you guys aware of any providers that do IP local loop service? What I want is to get a T-1 from said provider, plug it into my Cisco router, speak SIP to a voice gateway upstream, and have phone calls go out over PSTN from there. This is kind of what Vonage and ATT CallVantage do, but they are more geared toward the residential market, and I want to be able to bring an arbritary number of lines in. If you want local service, you have to tell us what is local to you, right? Care to finish the details so those on the list can help. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channelized T1, SIP phones, HW Echo Canceller
I have a channelized T1 coming in from our telco, terminated onto a TE405. There are three channelbanks serving internal analog extensions, and about 10 Cisco 7960s. I have no reports of echo on the analog extensions (as expected). The 7960 users complain of occasional echo (seems like 1 in 5 calls). Only the SIP user hears the echo, not the caller. I have echocancel=yes, echotraining=yes, echocancelwhenbridged=yes. Changes in the taps of echotraining have made things worse, so I have left it alone. I have backed the txgain down, as audio going out on the telco T1 is really hot. Even at -6dB gain, it is still notably louder from outside than other audio (comparing the ring generated by the telco when calling into asterisk with the ring generated by asterisk calling a station from the auto-attendant). If I drop gain to anything less that -6, I lose all audio. Would a hardware echo canceller deal with this type of echo? My understanding is that it is a result of sip being non-realtime and introducing latency (the latency being half the difference from the original utterance and the echo). Is this correct, or do I have it all wrong? From my studying of the list archives on this subject, it seems that there is no answer for Why is it so intermittent, other than to say that the problem originates somewhere in the two-wire system of the remote party. Is that correct? Has anyone heard of any kind of contraption to use just a single Tellabs card outside of the chassis? If possible, I'd like to avoid the cabling mess of a full tellabs chassis just to use one card. I have looked for a single-card chassis, but with no luck. Any pointers? Many thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with IAX , voice Distortion or Breakage.
Rich Adamson wrote: We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a robot's voice. This problem emerges whichever phone is called or whichever phone a call is made from (softphone, ipphone, local phones thru TDM cards) Without more technical data, its hard to guess. Might consider... 1. if you're using cisco phones, upgrade * to current Head or Stable on both ends of the iax link. The Stable code was just fixed on Friday. (There could be other phones/adapters that are impacted by the iax/gsm timestamp problems.) I've also noticed a fair number of other fixes that were just recently applied to Head. OK I will update asterisk. Yes we are using Cisco Phones. And simple phones connected to digium TDM cards. But problem was happend even with simple phones. 2. if not cisco phones, check the config's on whatever phone you're using to ensure transmit silence is enabled. If you are using the xten soft phone, the parameter is in the Advanced Settings area. 3. Check to ensure all ethernet nic adapters (and associated switch ports) are running in full-duplex mode, etc. 4. If none of the above apply, it would be helpful to see a packet trace (using ethereal) at about the time the distorting/failure is occurring. I have dump file, but it is about 3 Mb in archive. So, how I can send it to you? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Alexey Ostrovsky Sysadmin Ionidea UA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to get 48 volt Power Supplies for Cisco IP Phones
On Wed, 12 May 2004, Tom wrote: At 07:04 AM 5/12/2004, you wrote: Hi, I picked up some Cisco IP phones 7940, however, was foolish to not catch the fact that they do not come with power supplies.. Cisco power supplies for them are $150 (Can you believe it..) or more from a retailer I know. I found one place that sold compatible ones for $15 aus but with a 8 week turn around.. Can anyone point me in the direction where I can do some Mail Order of 48volt power supplies (240 AC in Australia) I would not buy the Cisco supply. I use the 3com 3cnjpse on most of our Cisco 7940g and 7960g phones. They are available in the US for less than $20 each. You need a custom wired Cat 5 cable (easy) from the power injector to the phone but this allows one less cord plugged into the back of the phone cluttering your desktop. http://www.voip-info.org/wiki-Cisco+POE That is a link to the instructions for making your own POE injector cable to use a standard 48v POE injector with a 79xx phone. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions/sip from database?
We are planning to deploy a pretty large asterisk server with many SIP extensions (might be up to 1 in the future), and I have a few questions: 1) is this possible, or are we running into some kind of limitation in the software that I wasn't aware of and that I didn't find by browsing through the archives and through Wiki? No, we don't need any G729-G711 transformations, it would only be acting as a SIP proxy (even if asterisk isn't a proxy). 2) is there a way to store extensions.conf and/or sip.conf in some kind of database, maybe MySQL? This would make life easier if someone wanted to change his SIP password. Or how would you otherwise solve this problem? 3) is there a quick way of reloading only a part of sip.conf/extensions.conf, for example if only a user password changed, or an extension's behaviour (eg. routing to voicemail instead of a SIP user)? Maybe I'm looking at the wrong software here and SER would be better for what I want to do... I know asterisk is supposed to be a PBX replacement, but the functions and flexibility it has really tells me stick with asterisk. Or am I way off with these assumptions? Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP local loop?
Yep, you are absolutely right. Sorry for the oversight :). Local to me is Dallas, TX. Shaun --- Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2004-05-24 at 12:19, Shaun Dawson wrote: Are you guys aware of any providers that do IP local loop service? What I want is to get a T-1 from said provider, plug it into my Cisco router, speak SIP to a voice gateway upstream, and have phone calls go out over PSTN from there. This is kind of what Vonage and ATT CallVantage do, but they are more geared toward the residential market, and I want to be able to bring an arbritary number of lines in. If you want local service, you have to tell us what is local to you, right? Care to finish the details so those on the list can help. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Domains Claim yours for only $14.70/year http://smallbusiness.promotions.yahoo.com/offer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
The problem is probably that both of your SIP phones are using the same port. I played with two 7960's behind a Linksys on Saturday and finally got them playing right when I changed the following: In Phone 1's SIP[macaddr].cnf: voip_control_port: 5061 In Phone 2's SIP[macaddr].cnf: voip_control_port: 5062 The default control port is 5060. Note: This is the port that the PHONE uses to initiate the connection to * and not the port it is connecting to. I'm having a similar problem with snom 200s would changing the port work there also or is that just a 7960 issue? Do you or any other know where I would change that on a snom 200 ?? thanks in advance Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI problem???
Yes, my termination point is a good 100' run away (cat5) from the server. I have experimented with settings of 0, 1, and 2 with no difference. I had only put the channel bank in a backup timing source (since removed) to see if there was any impact. I appreciate the libpri comment, I'll search the bug reports. [nothing remarkable in the logs, I have not yet decided to become a masochist and turn on PRI debugging on a production sysmem maybe if the pain level goes higher] Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Stingel Sent: Monday, May 24, 2004 12:58 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] PRI problem??? Hi Tim- Except for maybe eliminating the channel bank as a secondary source of timing, your conf files look fine. I notice that you've used 2 as the build-out (line length) parameter - I take it that your asterisk box is not right next to the switch. Anyway, I've never seen build-out make much difference, but maybe that's an issue. Although asterisk restarts B channels on a periodic basis, it's not supposed to do this if there's a call in progress, so there's something wrong. I noticed a problem recently that was causing D channels to drop under load, but after reporting this, Mark S fixed this on May 18th, so your more recent distr should cover it. Finally, in looking through libpri bugs on the bug list, Mark refers to some kind of firmware problem on some TE410P's (I think), so you might read through recent bug reports and see if any match your symptom (read the closed ones too) Is there anything remarkable in your /var/log/asterisk/messages log file? regards Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Timothy R. McKee Sent: Sunday, May 23, 2004 7:52 PM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] PRI problem??? I have just finished installing a new asterisk box at my work. The box is quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4 port Digium T1 card for channel bank and PRI access. I activated a PRI from a local CLEC (DMS-500 based, National protocol). This PRI is on slot 2 of the card and is set as the primary timing source. It is ESF/B8ZS. All the software is latest CVS HEAD as of 5/22/04. My problem lies in random intermittent drops of calls. The entire PRI seems to disappear, dropping all current established calls. I see occasional printouts on an asterisk management console showing all 23 B channels resetting with no reason given: configs: zaptel.conf # # span 1 is for channel bank (24 FXS) span=1,2,2,esf,b8zs # span 2 is for NuVox PRI (1-23B, 24D) span=2,1,2,esf,b8zs # span 3 is unused span=3,0,0,esf,b8zs # span 4 is unused span=4,0,0,esf,b8zs # # zapata.conf [channels] ; ; Default language ; language=en ; ; Default context ; context=dialnational ; signalling=fxo_ks use_callerid=yes callwaiting=no restrictcid=no usecallingpres=yes callwaitingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=yes relaxdtmf=no rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no amaflags=billing channels 1-24 omitted ; ; NuVox PRI context=sdnpri ; switchtype = national pridialplan = unknown signalling = pri_cpe callerid=asreceived group = 2 channel = 25-47 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to get 48 volt Power Supplies for Cisco IP Phones
Greg Boehnlein wrote: On Wed, 12 May 2004, Tom wrote: At 07:04 AM 5/12/2004, you wrote: Hi, I picked up some Cisco IP phones 7940, however, was foolish to not catch the fact that they do not come with power supplies.. Cisco power supplies for them are $150 (Can you believe it..) or more from a retailer I know. I found one place that sold compatible ones for $15 aus but with a 8 week turn around.. Can anyone point me in the direction where I can do some Mail Order of 48volt power supplies (240 AC in Australia) I would not buy the Cisco supply. I use the 3com 3cnjpse on most of our Cisco 7940g and 7960g phones. They are available in the US for less than $20 each. You need a custom wired Cat 5 cable (easy) from the power injector to the phone but this allows one less cord plugged into the back of the phone cluttering your desktop. http://www.voip-info.org/wiki-Cisco+POE That is a link to the instructions for making your own POE injector cable to use a standard 48v POE injector with a 79xx phone. I was just looking earlier and there are alot advertised on Ebay. Sub £20 is a good price. Shipped from Hong Kong usually tho. Nicholas Ruddick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
In sip.conf, try setting canreinvite=no for both lines. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 On Mon, 24 May 2004, Barry Fawthrop wrote: The problem is probably that both of your SIP phones are using the same port. I played with two 7960's behind a Linksys on Saturday and finally got them playing right when I changed the following: In Phone 1's SIP[macaddr].cnf: voip_control_port: 5061 In Phone 2's SIP[macaddr].cnf: voip_control_port: 5062 The default control port is 5060. Note: This is the port that the PHONE uses to initiate the connection to * and not the port it is connecting to. I'm having a similar problem with snom 200s would changing the port work there also or is that just a 7960 issue? Do you or any other know where I would change that on a snom 200 ?? thanks in advance Barry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
I'm having a similar problem with snom 200s would changing the port work there also or is that just a 7960 issue? Do you or any other know where I would change that on a snom 200 ?? thanks in advance Barry ___ try adding Canreinvite=no To UA in question inside sip.conf! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing multiple extensions
I've tried to setup multiple extension dialing - ie dial 1 number and it rings at a number of sources. For the most part its worked Now if someone dials 107 it rings Sip phones at 102 and 107, then goes to voicemail after 40 seconds. exten = 107,1,Dial(SIP/102SIP/107,40|r) exten = 107,2,Voicemail([EMAIL PROTECTED]) exten = 107,3,Hangup exten = 107,102,Voicemail([EMAIL PROTECTED]) exten = 107,103,Hangup The problem I'm running into is when I add my cell phone in exten = 107,1,Dial(SIP/102SIP/107Zap/2/11235551212,40|r) exten = 107,2,Voicemail([EMAIL PROTECTED]) exten = 107,3,Hangup exten = 107,102,Voicemail([EMAIL PROTECTED]) exten = 107,103,Hangup Calling from x110 the SIP extensions ring once, maybe twice. Then once my cell phone starts ringing the SIP phones stop, even though I haven't answered my cell phone. Here's what happens: -- Executing Dial(SIP/110-3e4f, SIP/107SIP/102Zap/2/11235551212|40|r) in new stack -- Called 107 -- Called 102 -- Called 2/11235551212 -- SIP/107-3d25 is ringing -- SIP/102-1f69 is ringing -- Zap/2-1 answered SIP/110-3e4f I know I can do the following exten = 107,1,Dial(SIP/102SIP/107,40|r) exten = 107,2,Dial(ZAP/2/11235551212,40|r) exten = 107,3,Voicemail([EMAIL PROTECTED]) exten = 107,4,Hangup exten = 107,102,Voicemail([EMAIL PROTECTED]) exten = 107,103,Hangup I'm just wondering if I could get all this in one line. Would dialing via IAX2 help rather then through the zaptel lines? -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor[EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 x102 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR destination when user presses '#'
If '#' is pressed during a call the CDR that is written at the end of the call contains '#' in the dst / destination field rather than the number that was originally called. How do I avoid losing that original number so that I can use the CDR for billing? I've tried not having a '#' target in extensions.conf and I've tried calling ResetCDR(w) in the '#' target hoping that would cause a CDR to be written with the original number but in both cases the CDR still contains '#'. Any ideas please? Thanks, Mark. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing multiple extensions
On Mon, 2004-05-24 at 13:47, Roger wrote: I've tried to setup multiple extension dialing - ie dial 1 number and it rings at a number of sources. The problem I'm running into is when I add my cell phone in exten = 107,1,Dial(SIP/102SIP/107Zap/2/11235551212,40|r) exten = 107,2,Voicemail([EMAIL PROTECTED]) exten = 107,3,Hangup exten = 107,102,Voicemail([EMAIL PROTECTED]) exten = 107,103,Hangup Calling from x110 the SIP extensions ring once, maybe twice. Then once my cell phone starts ringing the SIP phones stop, even though I haven't answered my cell phone. Here's what happens: -- Executing Dial(SIP/110-3e4f, SIP/107SIP/102Zap/2/11235551212|40|r) in new stack -- Called 107 -- Called 102 -- Called 2/11235551212 -- SIP/107-3d25 is ringing -- SIP/102-1f69 is ringing -- Zap/2-1 answered SIP/110-3e4f Your problem here probably lies in you are most likely using an analog line for the zap portion. Since analog doesn't know when the other side answered or not, it assumes answered and stops ringing your SIP phones. exten = 107,1,Dial(SIP/102SIP/107,40|r) exten = 107,2,Dial(ZAP/2/11235551212,40|r) exten = 107,3,Voicemail([EMAIL PROTECTED]) exten = 107,4,Hangup exten = 107,102,Voicemail([EMAIL PROTECTED]) exten = 107,103,Hangup I'm just wondering if I could get all this in one line. Would dialing via IAX2 help rather then through the zaptel lines? IAX2 would help, but only if you where using a provider who had a PRI or T1 line that knows when the other side answers the phone. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] dialing multiple extensions
You can use progress detect because on analog it will answer once its done dialing otherwise. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roger Sent: Monday, May 24, 2004 1:48 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dialing multiple extensions I've tried to setup multiple extension dialing - ie dial 1 number and it rings at a number of sources. For the most part its worked Now if someone dials 107 it rings Sip phones at 102 and 107, then goes to voicemail after 40 seconds. exten = 107,1,Dial(SIP/102SIP/107,40|r) exten = 107,2,Voicemail([EMAIL PROTECTED]) exten = 107,3,Hangup exten = 107,102,Voicemail([EMAIL PROTECTED]) exten = 107,103,Hangup The problem I'm running into is when I add my cell phone in exten = 107,1,Dial(SIP/102SIP/107Zap/2/11235551212,40|r) exten = 107,2,Voicemail([EMAIL PROTECTED]) exten = 107,3,Hangup exten = 107,102,Voicemail([EMAIL PROTECTED]) exten = 107,103,Hangup Calling from x110 the SIP extensions ring once, maybe twice. Then once my cell phone starts ringing the SIP phones stop, even though I haven't answered my cell phone. Here's what happens: -- Executing Dial(SIP/110-3e4f, SIP/107SIP/102Zap/2/11235551212|40|r) in new stack -- Called 107 -- Called 102 -- Called 2/11235551212 -- SIP/107-3d25 is ringing -- SIP/102-1f69 is ringing -- Zap/2-1 answered SIP/110-3e4f I know I can do the following exten = 107,1,Dial(SIP/102SIP/107,40|r) exten = 107,2,Dial(ZAP/2/11235551212,40|r) exten = 107,3,Voicemail([EMAIL PROTECTED]) exten = 107,4,Hangup exten = 107,102,Voicemail([EMAIL PROTECTED]) exten = 107,103,Hangup I'm just wondering if I could get all this in one line. Would dialing via IAX2 help rather then through the zaptel lines? -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor[EMAIL PROTECTED] Rockford, IL 61101 815-968-9888 x102 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fw: setting the number of rings befor asterisk picks up?
Hank-- I was waiting for the 4th, 5th, or 6th email to reply... BUT Have you looked at the Wait(seconds) application ? show application wait from the Asterisk CLI ? Maybe try that before you issue an Answer() on the line ? --Chris On Mon, 24 May 2004, hank waxed: - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. - Original Message - From: hank [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, May 23, 2004 6:19 PM Subject: setting the number of rings befor asterisk picks up? hello how do I set the number of rings picks up on? I am using a single port fxo card and currently asterisk is answering after 1 or 2 rings and I want it answering after 4 5 or 6 rings thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TerraCall Setting
Dear All, Any one know the correct SIP setting for the TerraCall? Thank You. Cary LEUNG ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: snom reporting busy when it shouldn't
Well, how many INVITE are being sent to the phone? If it receives the second one, it will report a busy. This sometimes happens when a user agents has a dangling registration and the SIP proxy does call forking. However, in this case the 2nd reject will not be propagated to the user agent client (RFC3261). A SIP trace of the phone will reveal this situation. CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of nicolas Sent: Monday, May 24, 2004 3:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE: snom reporting busy when it shouldn't I have no idea what is is, would be great if you can help me. I think this problem is in conjuction with the problem that when i dial the snoms in idle, without answering (exten == s,1,answer) before. Then it is ringing once and then * becoming an busy too. (May be if the 2. ring is coming). Thanks Christian nicolas Christian Stredicke wrote: Did you check if the phone is in DND state? Is there anything strange on the display? CS -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of nicolas Sent: Sunday, May 23, 2004 5:43 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] snom reporting busy when it shouldn't I am using asterisk cvs. Incoming/Outgoing calls are working. Calling the phone when some other lines are in use on the phone is ok. What does not work though is when the phone is ringing, nobody else can call the phone anymore. That's what * is saying: -- Got SIP response 486 Busy Here back from 192.168.1.250 -- SIP/snom1-4a44 is busy I am using the 2.05e snom200 firmware. Snom people sad must run. nico ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #3883 - 13 msgs
swar sir, can u please unsubscribe me for your list b.regards jihad chalhoub --- [EMAIL PROTECTED] wrote: Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Asterisk-oh323 0.6.1 Compiling problem (Michael Manousos) 2. Re: IP local loop? (Steven Critchfield) 3. Channelized T1, SIP phones, HW Echo Canceller (Steve Creel) 4. Re: Help with IAX , voice Distortion or Breakage. (Alexey Ostrovsky) 5. Re: Where to get 48 volt Power Supplies for Cisco IP Phones (Greg Boehnlein) 6. extensions/sip from database? (Manuel Wenger) 7. Re: IP local loop? (Shaun Dawson) 8. Re: 2 Sip phones behind un-natted Asterisk (Barry Fawthrop) 9. RE: PRI problem??? (Timothy R. McKee) 10. Re: Where to get 48 volt Power Supplies for Cisco IP Phones (Nicholas Ruddick) 11. Re: 2 Sip phones behind un-natted Asterisk (Bruce Komito) --__--__-- Message: 1 Date: Mon, 24 May 2004 20:32:05 +0300 From: Michael Manousos [EMAIL PROTECTED] Organization: inAccess Networks To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk-oh323 0.6.1 Compiling problem Reply-To: [EMAIL PROTECTED] I need the full output for this (the first lines are missing). Michael. Nicholas Ruddick wrote: ok done, but now i'm getting different errors - /usr/src/pwlib/include/ptlib/args.h:389: virtual outside class declaration /usr/src/pwlib/include/ptlib/args.h:389: non-member function `UnknownOption (...)' cannot have `const' method qualifier [snip...] in this scope /usr/src/pwlib/include/ptlib/indchan.h:259: `readChannel' was not declared in this scope /usr/src/pwlib/include/ptlib/indchan.h:261: `PChannel' was not declared in this scope /usr/src/pwlib/include/ptlib/indchan.h:261: `writeChannel' was not declared in this scope /usr/src/pwlib/include/ptlib/indchan.h:263: parse error before `=' /usr/src/pwlib/include/ptlib/indchan.h:265: `BOOL Open (...)' redeclared as different kind of symbol /usr/src/pwlib/include/ptlib/indchan.h:229: previous declaration of `BOOL Open' /usr/src/pwlib/include/ptlib/indchan.h:229: previous non-function declaration `BOOL Open' /usr/src/pwlib/include/ptlib/indchan.h:265: conflicts with function declaration `BOOL Open (...)' /usr/src/pwlib/include/ptlib/indchan.h:265: confused by earlier errors, bailing out make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.1/wrapper' make: *** [subdirs_all] Error 1 Whats this all about, it's still complaining about some audio thing i just can't work out. I'm using redhat 7.3 btw, i have both the openh323, pwlib standard, devel and src packages install. Still no joy. Thanks, Nicholas Ruddick Pablo Endres wrote: Check your README file again. In order to compile 0.6.1 you need newer versions of pwlib and openh323 (1.6.6 and 1.13.5) Then it should work just fine Pablo --__--__-- Message: 2 Subject: Re: [Asterisk-Users] IP local loop? From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Mon, 24 May 2004 12:32:12 -0500 Reply-To: [EMAIL PROTECTED] On Mon, 2004-05-24 at 12:19, Shaun Dawson wrote: Are you guys aware of any providers that do IP local loop service? What I want is to get a T-1 from said provider, plug it into my Cisco router, speak SIP to a voice gateway upstream, and have phone calls go out over PSTN from there. This is kind of what Vonage and ATT CallVantage do, but they are more geared toward the residential market, and I want to be able to bring an arbritary number of lines in. If you want local service, you have to tell us what is local to you, right? Care to finish the details so those on the list can help. -- Steven Critchfield [EMAIL PROTECTED] --__--__-- Message: 3 Date: Mon, 24 May 2004 13:34:02 -0400 (EDT) From: Steve Creel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Channelized T1, SIP phones, HW Echo Canceller Reply-To: [EMAIL PROTECTED] I have a channelized T1 coming in from our telco, terminated onto a TE405. There are three channelbanks serving internal analog extensions, and about 10 Cisco 7960s. I have no reports of echo on the analog extensions (as expected). The 7960 users complain of occasional echo (seems like 1 in 5 calls). Only the SIP user hears the echo, not the caller. I have echocancel=yes, echotraining=yes,
Re: [Asterisk-Users] dialing multiple extensions
I'm just wondering if I could get all this in one line. Would dialing via IAX2 help rather then through the zaptel lines? I have also seen key systems before that will ring your cell phone and prompt you to press 1 if you would like to accept the call, or press 2 if you would like to enter another number. It may take a lot of programming, but you could write maybe an AGI script to not consider the Zaptel line answered (even if it is analog) to be considered answered unless it receives the DTMF tone of 1. This would be beneficial in cases where you have Voicemail on your cell phone, so the call wasn't considered answered when your Voicemail picks it up. Just a thought.. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Prepaid
Hi Steve, Sounds like more or less what I want. I would be greatful if you could send me your patch. Just wondering if you play any prompts to the user at all ? like when the credit is running out etc. Thanks Umar. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen Davies Sent: 24 May 2004 07:37 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Prepaid On Mon, 24 May 2004, usedcanon wrote: I have a requirement for a setup with prepaid call credits. I am aware of the two applications available (been researching for the past week), app_prepaid and app_rateengine. However neither of the two sound like exactly what I want. However I was wondering that someone who has used it might be able to say if they could be used in my scenario. Basically my scenario is pretty straight forward. Credit will be allocated to the ddi, I dont need any announcements etc (maybe low credit warning during call could be useful thoug). From the users prespective everything will be transparent. However the call should disconnect when the credit runs out. The CDR and the account DB need to be adjusted according to the call made. My guess is that app_prepaid could used with modification, I am assuming here that this is not possible as-is with configuration. Basically in case of the prepaid app, the card number can be replace transparently with the callerID. Hi, I did this to app_prepaid - you can pass a parameter into Prepaid() - its looked up in a table to find an associated card number - if that is found then the card number prompt is skipped and the associated card is used automatically. I can send a patch if you like (will also include a minor change or two to have app_prepaid work against CVS. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TerraCall Setting
I posted them for you yesterday. [EMAIL PROTECTED] wrote: Dear All, Any one know the correct SIP setting for the TerraCall? Thank You. Cary LEUNG ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dialing multiple extensions
On Mon, 2004-05-24 at 13:24, Brent Franks wrote: I'm just wondering if I could get all this in one line. Would dialing via IAX2 help rather then through the zaptel lines? I have also seen key systems before that will ring your cell phone and prompt you to press 1 if you would like to accept the call, or press 2 if you would like to enter another number. It may take a lot of programming, but you could write maybe an AGI script to not consider the Zaptel line answered (even if it is analog) to be considered answered unless it receives the DTMF tone of 1. This would be beneficial in cases where you have Voicemail on your cell phone, so the call wasn't considered answered when your Voicemail picks it up. Seems there used to be an option to dial that did that. I think the called party had to press # to signify an answer. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] testing asterisk on FXS lines
I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to incoming so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten = s,1,Wait(1) exten = s,2,Answer() exten = s,3,Background(hello2) ; this is the file I need to test the playback of first And I do a restart. When I pickup one of the FXS handsets, though, I get this from asterisk (running with the -vvvc arg): Starting simple switch on 'Zap/1-1' and that is it. I know that the context is right because I put a hard-dial of 202 in there and when I dialed it, it would connect to that extension (Zap/2) and if I dialed anything else I would get fast busy. I have checked and the line right after the last exten above is another context marker. The asterisk output also shows the s extensions being loaded under the correct context when I do a reload after the restart (to see just the messages from the contexts being loaded). What am I missing to get the FXS lines, in the context incoming, to do the wait/answer/background? Thanks! -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NCS support?
Does anyone know if there is a version of Asterisk that supports the PacketCable NCS standard (flavor of mgcp). Thanks. Bob Naylor Brix Networks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #3883 - 13 msgs
there is info at the bottum of each and every message that is sent to this list please read info to unsubscribe. hth hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. - Original Message - From: jihad chalhoub [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, May 24, 2004 12:29 PM Subject: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #3883 - 13 msgs swar sir, can u please unsubscribe me for your list b.regards jihad chalhoub --- [EMAIL PROTECTED] wrote: Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Asterisk-oh323 0.6.1 Compiling problem (Michael Manousos) 2. Re: IP local loop? (Steven Critchfield) 3. Channelized T1, SIP phones, HW Echo Canceller (Steve Creel) 4. Re: Help with IAX , voice Distortion or Breakage. (Alexey Ostrovsky) 5. Re: Where to get 48 volt Power Supplies for Cisco IP Phones (Greg Boehnlein) 6. extensions/sip from database? (Manuel Wenger) 7. Re: IP local loop? (Shaun Dawson) 8. Re: 2 Sip phones behind un-natted Asterisk (Barry Fawthrop) 9. RE: PRI problem??? (Timothy R. McKee) 10. Re: Where to get 48 volt Power Supplies for Cisco IP Phones (Nicholas Ruddick) 11. Re: 2 Sip phones behind un-natted Asterisk (Bruce Komito) --__--__-- Message: 1 Date: Mon, 24 May 2004 20:32:05 +0300 From: Michael Manousos [EMAIL PROTECTED] Organization: inAccess Networks To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk-oh323 0.6.1 Compiling problem Reply-To: [EMAIL PROTECTED] I need the full output for this (the first lines are missing). Michael. Nicholas Ruddick wrote: ok done, but now i'm getting different errors - /usr/src/pwlib/include/ptlib/args.h:389: virtual outside class declaration /usr/src/pwlib/include/ptlib/args.h:389: non-member function `UnknownOption (...)' cannot have `const' method qualifier [snip...] in this scope /usr/src/pwlib/include/ptlib/indchan.h:259: `readChannel' was not declared in this scope /usr/src/pwlib/include/ptlib/indchan.h:261: `PChannel' was not declared in this scope /usr/src/pwlib/include/ptlib/indchan.h:261: `writeChannel' was not declared in this scope /usr/src/pwlib/include/ptlib/indchan.h:263: parse error before `=' /usr/src/pwlib/include/ptlib/indchan.h:265: `BOOL Open (...)' redeclared as different kind of symbol /usr/src/pwlib/include/ptlib/indchan.h:229: previous declaration of `BOOL Open' /usr/src/pwlib/include/ptlib/indchan.h:229: previous non-function declaration `BOOL Open' /usr/src/pwlib/include/ptlib/indchan.h:265: conflicts with function declaration `BOOL Open (...)' /usr/src/pwlib/include/ptlib/indchan.h:265: confused by earlier errors, bailing out make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.6.1/wrapper' make: *** [subdirs_all] Error 1 Whats this all about, it's still complaining about some audio thing i just can't work out. I'm using redhat 7.3 btw, i have both the openh323, pwlib standard, devel and src packages install. Still no joy. Thanks, Nicholas Ruddick Pablo Endres wrote: Check your README file again. In order to compile 0.6.1 you need newer versions of pwlib and openh323 (1.6.6 and 1.13.5) Then it should work just fine Pablo --__--__-- Message: 2 Subject: Re: [Asterisk-Users] IP local loop? From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Mon, 24 May 2004 12:32:12 -0500 Reply-To: [EMAIL PROTECTED] On Mon, 2004-05-24 at 12:19, Shaun Dawson wrote: Are you guys aware of any providers that do IP local loop service? What I want is to get a T-1 from said provider, plug it into my Cisco router, speak SIP to a voice gateway upstream, and have phone calls go out over PSTN from there. This is kind of what Vonage and ATT CallVantage do, but they are more geared toward the residential market, and I want to be able to bring an arbritary number of lines in. If you want local service, you have to tell us what is local to you, right? Care to finish the details so those on
[Asterisk-Users] using the asterisk mailbox utility
hello according to this user guide found at http://www.automated.it/guidetoasterisk.htm#_Toc49248768 it says the following Voicemail - Please leave a message after the tone... Ok, so you've got the basics going, and it's great - if you happen to sit by you phone all the time. What happens if you are out/away from your desk/sleeping you'll miss those vital calls. We need to set up voicemail to capture all those messages if we miss them. The first thing we need to do is create the mailbox for Asterisk to use, thankfully there is a little utility to do this: /usr/src/asterisk/addmailbox You'll be prompted for a mailbox number, when I type in the following command /usr/src/asterisk/addmailbox it says that the command can't be found is this utility still in asterisk? or is this user guide in correct on this part of the set up? so far these user guides are verry self explanitory and I am able to under stand them quite well thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. time is the fire in which we burn, Tollian Soran. grudges aren't worth holding--One who holds them shows his self-weakness. Contact info: [EMAIL PROTECTED] Email: Same as MSN. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323, video and asterisk....
Just few question about H.323 1) I can authenticate H.323 users without a Gatekeeper 2) If i have two asterisk server, connected with an IAX2 trunk, an H.323 client on the server 1 can make video call to an H.323 client on server 2 ??? 3) An H.323 client can make a video conference with a SIP client (with video capabilities...) ??? Igor ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using the asterisk mailbox utility
quote who=hank
hello according to this user guide found at
http://www.automated.it/guidetoasterisk.htm#_Toc49248768
it says the following
Voicemail - Please leave a message after the tone...
Ok, so you've got the basics going, and it's great - if you happen to sit by
you phone all the time. What happens if you are out/away from your
desk/sleeping
you'll miss those vital calls. We need to set up voicemail to capture all
those messages if we miss them.
edit /etc/asterisk/voicemail.conf
Then you will need to edit /etc/asterisk/extensions.conf
I use a macro for all my extensions:
[macro-stdextn]
exten = s,1,Dial(${ARG2},20,t)
exten = s,2,VoiceMail2(u${ARG1})
exten = s,3,Hangup
exten = s,102,VoiceMail2(b${ARG1})
exten = s,103,Hangup
--
END OF LINE
-MCP
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Re: [Asterisk-Users] using the asterisk mailbox utility
Hi! The first thing we need to do is create the mailbox for Asterisk to use, thankfully there is a little utility to do this: /usr/src/asterisk/addmailbox That's very old stuff, not needed anymore. is this utility still in asterisk? or is this user guide in correct on this part of the set up? Read more here: http://www.voip-info.org/wiki-Asterisk+VoiceMail Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] testing asterisk on FXS lines
On May 24, 2004, at 4:00 PM, Michael George wrote: I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to incoming so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten = s,1,Wait(1) exten = s,2,Answer() exten = s,3,Background(hello2) ; this is the file I need to test the playback of first And I do a restart. When I pickup one of the FXS handsets, though, I get this from asterisk (running with the -vvvc arg): Starting simple switch on 'Zap/1-1' and that is it. I know that the context is right because I put a hard-dial of 202 in there and when I dialed it, it would connect to that extension (Zap/2) and if I dialed anything else I would get fast busy. I have checked and the line right after the last exten above is another context marker. The asterisk output also shows the s extensions being loaded under the correct context when I do a reload after the restart (to see just the messages from the contexts being loaded). What am I missing to get the FXS lines, in the context incoming, to do the wait/answer/background? Thanks! For some reason, the s extension is not being executed for the FXS lines. I changed their default context back to internal and added exten = s,1,Background(hello2) to the internal context, thinking that when I pick up the handset I will get the hello2 audio file played as it waits for me to enter digits. But the audio file is not played... I must be missing an essential concept here... -Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Making a SIP call
I am still having this problem of only capturing part of the IP address, I am currently checking into a possible hardware/software issue on the client side but was wondering if there are any setting I need to set on the asterisk server to allow an peer to peer call. I have set dtmfmode=inband. Is there anything else I need to set? Brian Message: 5 From: David J Carter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Making a SIP call Date: Sat, 22 May 2004 00:14:55 +0100 Reply-To: [EMAIL PROTECTED] Check your sip.conf Make sure the dtmfmode is set the same as the phone. I had this before. Usually to dial an IP address you have a keystroke before you enter the address. I think on a Grandstream phone you press the menu button then the IP address. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 21 May 2004 21:57 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Making a SIP call If someone could point me in the right direction I would much appreciate it. Here is my problem: My directions for my sip phone says to dial an ip address 12*34*65*78#. When I dial that into my phone my asterisk server is only picking up some of the numbers in the above example it would pick up 6578. Then of course not find it and ring busy on the phone. The same is true for dialing a regular phone number ( it seems to pick up 4 digits or so) I very new to setting this up so I imagine I need to make a change to a config file, but don't know where to start. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip Registration Problem
Title: Message Hi All, I had an unusual problem today; I'm sure it's a configuration problem. I had 2 phones behind a nat device and I had qualify=300 in both extensions config. The device I was talking to was an edgewater traffic shaper,/Sip Proxy. Since it is acting as a sip proxy, it was ignoring the OPTIONS messages that * was sending, and thus * interpreted that as the extensions being down. I removed the qualify lines and sip reload [ed]. The extension still showed up as "UNREACHABLE" instead of "UNMONITORED". I had to do a full restart to get it to stop sending the OPTIONS messages. What did I do wrong here? How can I make a change to qualify without restarting? Thanks all, Brett
Re: [Asterisk-Users] Re: Making a SIP call
[EMAIL PROTECTED] wrote: I am still having this problem of only capturing part of the IP address, I am currently checking into a possible hardware/software issue on the client side but was wondering if there are any setting I need to set on the asterisk server to allow an peer to peer call. I have set dtmfmode=inband. Is there anything else I need to set? dtmfmode=inband only works with the ulaw and alaw codecs. If you use any other codec you MUST use rfc2833 or info DTMF modes (set on the phone AND on Asterisk) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
