Re: [asterisk-users] REFER mesage extraction using SIP_HEADER
On Dec 1, 2007, at 12:30 AM, [EMAIL PROTECTED] wrote: > I would like to extract the information present in the SIP REFER > message that comes to asterisk. Would SIP_HEADER() allow me to do that > ? I have used SIP_HEADER() for extracting the to and from SIP headers > previously. I wanted to do the exact same thing a while ago. However it is not possible as far as I can tell. I've tried it and verified the headers are being sent, but asterisk can't see them. It can read the headers from the original INVITE. This bug report: http://bugs.digium.com/view.php?id=4934 Complained of the same thing, but was ended as too much work and folks weren't sure it's even correct. Then this one: http://bugs.digium.com/print_bug_page.php?bug_id=8378 Talks about the Refered-By header in REFER messages, which seems to have been folded in to 1.4. It didn't solve the general case of other headers, however. I worked around it in my case, since the original invite actually had what I needed most of the time. Some odd cases just will remain broken. Norman Franke ASD, Inc. www.myasd.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
Salvatore Giudice wrote: > They are cheap. You only have to pay for the box and the > maintenance percentage. That is indeed the Avaya way. First you buy it, then you rent it. Stop paying their maintenance fees and their dial into your PBX and cripple the OS by removing customer maintenance command permissions. > Hell, Avaya won't even > give you root on any of their servers. You cant audit the box and you can't > poll them unless you pay them money to join their partner program and get > their SDK. If you already have Avaya, you should just buy Message Networking > or a Mitel voicemail server if you want seamless voicemail with Avaya. > > However, you should know that using Avaya is probably a bad idea to begin > with. Until February 07, the majority Avaya's soft switch products were > running on Redhat 9, which was unsupported since 2003. Avaya was only > managing a dozen packages and they've always left it up to the customer to > know when they need an update, requiring the customer to request a field > load. It has to be the worst update model in the industry when it comes to > infrastructure monitoring and patching. By using Avaya, you are blindly > trusting them to properly maintain a Linux appliance. This is something they > are not capable of and you can't even audit them. > > Avaya is what happens to organizations when they have ignorant telecom > infrastructure engineers deciding what products to buy. Avaya focuses sales > on those engineers because they k now their products won't pass > certification by network, systems, or security engineers. Telecom engineers > only look for features and usually get their asses handed to them after they > put Avaya VoIP into their infrastructure. > Bravo. A well-deserved lambasting of this awful vendor. -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk & Cisco calling Name
Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling forward unconditional then Asterisk accepts the call but without cname. Below is a trace. Any help is appreciated. Thanks John Bittner Simlab.net voippbx01*CLI> <-- SIP read from 216.86.35.24:63549: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag="tgrp:PRI-TRUNK-GROUP1";branch=z9hG4bK111A56 From: ;tag=4F9EF08-163B To: Date: Sat, 01 Dec 2007 05:23:25 GMT Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer,replaces Min-SE: 1800 Cisco-Guid: 1613584196-2667844060-2152857615-892193345 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: "pending" ;party=calling;screen=yes;privacy=off Timestamp: 1196486605 Contact: Expires: 180 Allow-Events: telephone-event MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=uniqueBoundary Content-Length: 680 --uniqueBoundary Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 6852 2375 IN IP4 216.86.35.24 s=SIP Call c=IN IP4 216.86.35.24 t=0 0 m=audio 18472 RTP/AVP 0 101 c=IN IP4 216.86.35.24 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,NI***, USI,rate,c,s,c,1 USI,lay1,ulaw TMR,00 CPN,04,,1,9734333001 CGN,04,,1,y,4,9733901090 CPC,09 FCI,,,y, GCI,602d57449f0411dc8052000f352dca41 UFC,GEN,5,gentf,79 UFC,GEN,5,fachd,9f8b0100 UFC,GEN,5,inpdu,02010106072a8648ce150004 --uniqueBoundary-- --- (21 headers 33 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 216.86.35.24 : 5060 (non-NAT) Found peer '216.86.35.24' Transmitting (no NAT) to 216.86.35.24:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag="tgrp:PRI-TRUNK-GROUP1";branch=z9hG4bK111A56;received=216.86.35.24 From: ;tag=4F9EF08-163B To: ;tag=as39c359be Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: SimlabVOIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' voippbx01*CLI> <-- SIP read from 216.86.35.24:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag="tgrp:PRI-TRUNK-GROUP1";branch=z9hG4bK111A56 From: ;tag=4F9EF08-163B To: ;tag=as39c359be Date: Sat, 01 Dec 2007 05:23:25 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration state: Failed
Hi, I am using the OS which bundled with AsteriskNow - Original Message - From: Vivek Shrivastava To: Newbie Cc: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December 01, 2007 12:25 PM Subject: Re: [asterisk-users] Registration state: Failed Hmmm, what OS you are using,,,this could be related to "Access Control Lists"..but i guess that is in Solaris On 11/30/07, Newbie <[EMAIL PROTECTED]> wrote: Hello, After I turned on "full=>" in logged.conf .. I got the following: [Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 ' failed for ' 172.16.1.169' - Device does not match ACL [Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < sip:[EMAIL PROTECTED]>' failed for ' 172.16.1.169' - Device does not match ACL [Nov 30 12:00:33] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < sip:[EMAIL PROTECTED]>' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:[EMAIL PROTECTED]>' failed for ' 172.16.1.169' - Device does not match ACL [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:[EMAIL PROTECTED]>' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:36] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < sip:[EMAIL PROTECTED]>' failed for ' 172.16.1.169' - Device does not match ACL any idea or clue? Thanks a lot in advance Regards Winanjaya - Original Message - From: Vivek Shrivastava To: Newbie Cc: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December 01, 2007 11:50 AM Subject: Re: [asterisk-users] Registration state: Failed well, then i would recommend to see "full" log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with "full=>" in the logger.conf... the log will be the usual /var/log/asterisk/ directory. Thanks, Vivek On 11/30/07, Newbie <[EMAIL PROTECTED] > wrote: Hi, there is no problem with X-Lite, the problem is SPA-3102 shown: Line 1: Registration Status: Failed PSTN Line 1: Registration Status: Failed I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow.. since I am very new with this.. I don't know why this problem occurs ... could any body please help? Thanks & Regards Winanjaya [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm allow=ulaw allow=h261 allow=h263 allow=h263p register=998:[EMAIL PROTECTED]/998 register=999:[EMAIL PROTECTED]/999 [line1] type=peer host=dynamic defaultip=172.16.1.74 fromuser=998 secret=1234 fromdomain=172.16.1.169 [line2] type=peer host=dynamic defaultip=172.16.1.74 username=999 secret=1234 fromdomain=172.16.1.169 Command> sip show peers Name/username HostDyn Nat ACL Port Status pstnline1/999 (Unspecified)D 0Unmonitored line1 (Unspecified)D 0Unmonitored 250/250172.16.1.88 D 27778Unmonitored 2500 (Unspecified)D 0Unmonitored 251(Unspecified)D 0Unmonitored 5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] - Original Message - From: Vivek Shrivastava To: Newbie ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December 01, 2007 11:34 AM Subject: Re: [asterisk-users] Registration state: Failed Hi, x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues. Thanks, Vivek On 11/30/07, Newbie <[EMAIL PROTECTED] > wrote: Dear Support, I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line. I have 3 extensions: 250 -> my extension 998 -> I configured as Line 1 in SPA-3102
Re: [asterisk-users] Registration state: Failed
you can also look at this... http://www.asteriskguru.com/tutorials/idefisk_20_free.html "I has this error initially with Asterisk server when I try to register. " Device does not match ACL " got it resolved by setting Caller ID Name : " users exten " On 11/30/07, Vivek Shrivastava <[EMAIL PROTECTED]> wrote: > > Hmmm, what OS you are using,,,this could be related to "*Access Control > Lists"..*but i guess that is in Solaris * * > > On 11/30/07, Newbie <[EMAIL PROTECTED]> wrote: > > > > Hello, > > > > After I turned on "full=>" in logged.conf .. I got the following: > > > > [Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < > > sip:[EMAIL PROTECTED]>' failed for ' 172.16.1.169' - Device does not match > > ACL > > [Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < > > sip:[EMAIL PROTECTED]>' failed for ' 172.16.1.169' - Device does not match > > ACL > > [Nov 30 12:00:33] NOTICE[2601] chan_sip.c: Registration from 'FXS1 > > ' > > failed for '172.16.1.169' - Device does not match ACL > > [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < > > sip:[EMAIL PROTECTED]>' failed for ' 172.16.1.169' - Device does not match > > ACL > > [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 > > ' > > failed for '172.16.1.169' - Device does not match ACL > > [Nov 30 12:00:36] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < > > sip:[EMAIL PROTECTED]>' failed for ' 172.16.1.169' - Device does not match > > ACL > > > > any idea or clue? > > > > Thanks a lot in advance > > > > Regards > > > > Winanjaya > > > > > > > > - Original Message - > > *From:* Vivek Shrivastava <[EMAIL PROTECTED]> > > *To:* Newbie <[EMAIL PROTECTED]> > > *Cc:* Asterisk Users Mailing List - Non-Commercial Discussion > > > > *Sent:* Saturday, December 01, 2007 11:50 AM > > *Subject:* Re: [asterisk-users] Registration state: Failed > > > > > > well, then i would recommend to see "full" log in debug mode that might > > give some clue. if you have not done this before you can uncomment line > > starting with "full=>" in the logger.conf... the log will be the usual > > /var/log/asterisk/ directory. > > > > Thanks, > > > > Vivek > > > > > > On 11/30/07, Newbie <[EMAIL PROTECTED] > wrote: > > > > > > Hi, > > > there is no problem with X-Lite, the problem is SPA-3102 shown: > > > > > > Line 1: > > > Registration Status: Failed > > > > > > PSTN Line 1: > > > Registration Status: Failed > > > > > > I also had added 1 more extension 251..then tried to call 251 from 250 > > > by using X-Lite and it works perfectly.. so that's why I am sure there is > > > no > > > problem with X-Lite .. what I suspect is the problem on Registration > > > process > > > in AsteriskNow.. > > > > > > since I am very new with this.. I don't know why this problem occurs > > > ... could any body please help? > > > > > > Thanks & Regards > > > Winanjaya > > > > > > [general] > > > context=default > > > allowoverlap=no > > > bindport=5060 > > > bindaddr=0.0.0.0 > > > srvlookup=yes > > > videosupport=yes > > > disallow=all > > > allow=ilbc > > > allow=gsm > > > allow=ulaw > > > allow=h261 > > > allow=h263 > > > allow=h263p > > > register=998:[EMAIL PROTECTED]/998 > > > register=999:[EMAIL PROTECTED]/999 > > > [line1] > > > type=peer > > > host=dynamic > > > defaultip=172.16.1.74 > > > fromuser=998 > > > secret=1234 > > > fromdomain=172.16.1.169 > > > > > > [line2] > > > type=peer > > > host=dynamic > > > defaultip=172.16.1.74 > > > username=999 > > > secret=1234 > > > fromdomain=172.16.1.169 > > > > > > > > > Command>* sip show peers* > > > > > > Name/username HostDyn Nat ACL Port Status > > > pstnline1/999 (Unspecified)D 0 > > > Unmonitored > > > line1 (Unspecified)D 0 > > > Unmonitored > > > 250/250172.16.1.88 D 27778 > > > Unmonitored > > > 2500 (Unspecified)D 0 > > > Unmonitored > > > 251(Unspecified)D 0 > > > Unmonitored > > > 5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 > > > offline] > > > > > > > > > > > > > > > > > > > > > > > > > > > - Original Message - > > > > > > *From:* Vivek Shrivastava <[EMAIL PROTECTED]> > > > *To:* Newbie <[EMAIL PROTECTED]> ; Asterisk Users Mailing List - > > > Non-Commercial Discussion > > > *Sent:* Saturday, December 01, 2007 11:34 AM > > > *Subject:* Re: [asterisk-users] Registration state: Failed > > > > > > > > > Hi, > > > > > > x-lite has extensive debug facility you can turn that on in the > > > advanced options, that probably will give better understanding as what is > > > going on from x-lite side. i also have experienced the same but that > > > involved firewall and NAT issues. > > > > > > Thanks, > > > > > > Vivek > > > > > > > > > On 11/30/07, Newbie <[EMAIL PROTECTED] > wrote: > > > > > > > >
Re: [asterisk-users] Registration state: Failed
Hmmm, what OS you are using,,,this could be related to "*Access Control Lists"..*but i guess that is in Solaris * * On 11/30/07, Newbie <[EMAIL PROTECTED]> wrote: > > Hello, > > After I turned on "full=>" in logged.conf .. I got the following: > > [Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < > sip:[EMAIL PROTECTED]>' failed for '172.16.1.169' - Device does not match > ACL > [Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < > sip:[EMAIL PROTECTED]>' failed for '172.16.1.169' - Device does not match > ACL > [Nov 30 12:00:33] NOTICE[2601] chan_sip.c: Registration from 'FXS1 < > sip:[EMAIL PROTECTED]>' failed for '172.16.1.169' - Device does not match > ACL > [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < > sip:[EMAIL PROTECTED]>' failed for '172.16.1.169' - Device does not match > ACL > [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < > sip:[EMAIL PROTECTED]>' failed for '172.16.1.169' - Device does not match > ACL > [Nov 30 12:00:36] NOTICE[2601] chan_sip.c: Registration from 'FXO1 < > sip:[EMAIL PROTECTED]>' failed for '172.16.1.169' - Device does not match > ACL > > any idea or clue? > > Thanks a lot in advance > > Regards > > Winanjaya > > > > - Original Message - > *From:* Vivek Shrivastava <[EMAIL PROTECTED]> > *To:* Newbie <[EMAIL PROTECTED]> > *Cc:* Asterisk Users Mailing List - Non-Commercial > Discussion > *Sent:* Saturday, December 01, 2007 11:50 AM > *Subject:* Re: [asterisk-users] Registration state: Failed > > > well, then i would recommend to see "full" log in debug mode that might > give some clue. if you have not done this before you can uncomment line > starting with "full=>" in the logger.conf... the log will be the usual > /var/log/asterisk/ directory. > > Thanks, > > Vivek > > > On 11/30/07, Newbie <[EMAIL PROTECTED]> wrote: > > > > Hi, > > there is no problem with X-Lite, the problem is SPA-3102 shown: > > > > Line 1: > > Registration Status: Failed > > > > PSTN Line 1: > > Registration Status: Failed > > > > I also had added 1 more extension 251..then tried to call 251 from 250 > > by using X-Lite and it works perfectly.. so that's why I am sure there is no > > problem with X-Lite .. what I suspect is the problem on Registration process > > in AsteriskNow.. > > > > since I am very new with this.. I don't know why this problem occurs ... > > could any body please help? > > > > Thanks & Regards > > Winanjaya > > > > [general] > > context=default > > allowoverlap=no > > bindport=5060 > > bindaddr=0.0.0.0 > > srvlookup=yes > > videosupport=yes > > disallow=all > > allow=ilbc > > allow=gsm > > allow=ulaw > > allow=h261 > > allow=h263 > > allow=h263p > > register=998:[EMAIL PROTECTED]/998 > > register=999:[EMAIL PROTECTED]/999 > > [line1] > > type=peer > > host=dynamic > > defaultip=172.16.1.74 > > fromuser=998 > > secret=1234 > > fromdomain=172.16.1.169 > > > > [line2] > > type=peer > > host=dynamic > > defaultip=172.16.1.74 > > username=999 > > secret=1234 > > fromdomain=172.16.1.169 > > > > > > Command>* sip show peers* > > > > Name/username HostDyn Nat ACL Port Status > > pstnline1/999 (Unspecified)D 0Unmonitored > > line1 (Unspecified)D 0Unmonitored > > 250/250172.16.1.88 D 27778Unmonitored > > 2500 (Unspecified)D 0Unmonitored > > 251(Unspecified)D 0Unmonitored > > 5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 > > offline] > > > > > > > > > > > > > > > > > > - Original Message - > > > > *From:* Vivek Shrivastava <[EMAIL PROTECTED]> > > *To:* Newbie <[EMAIL PROTECTED]> ; Asterisk Users Mailing List - > > Non-Commercial Discussion > > *Sent:* Saturday, December 01, 2007 11:34 AM > > *Subject:* Re: [asterisk-users] Registration state: Failed > > > > > > Hi, > > > > x-lite has extensive debug facility you can turn that on in the advanced > > options, that probably will give better understanding as what is going on > > from x-lite side. i also have experienced the same but that involved > > firewall and NAT issues. > > > > Thanks, > > > > Vivek > > > > > > On 11/30/07, Newbie <[EMAIL PROTECTED] > wrote: > > > > > > Dear Support, > > > > > > I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 > > > connected with PSTN line. > > > > > > I have 3 extensions: > > > > > > 250 -> my extension > > > 998 -> I configured as Line 1 in SPA-3102 > > > 999 -> I configured as PSTN Line 1 in SPA-3102 > > > > > > I have created 998 and 999 to the user extension list of the > > > AsteriskNow > > > > > > why I still got Registration state: Failed for both Line 1 status and > > > PSTN Line status ? > > > > > > > > > my topology is below: > > > > > > Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line > > > > > > Please help > > > > > > Thanks
[asterisk-users] REFER mesage extraction using SIP_HEADER
Hi * users, I would like to extract the information present in the SIP REFER message that comes to asterisk. Would SIP_HEADER() allow me to do that ? I have used SIP_HEADER() for extracting the to and from SIP headers previously. Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration state: Failed
Hello, After I turned on "full=>" in logged.conf .. I got the following: [Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 ' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 ' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:33] NOTICE[2601] chan_sip.c: Registration from 'FXS1 ' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 ' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 ' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:36] NOTICE[2601] chan_sip.c: Registration from 'FXO1 ' failed for '172.16.1.169' - Device does not match ACL any idea or clue? Thanks a lot in advance Regards Winanjaya - Original Message - From: Vivek Shrivastava To: Newbie Cc: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December 01, 2007 11:50 AM Subject: Re: [asterisk-users] Registration state: Failed well, then i would recommend to see "full" log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with "full=>" in the logger.conf... the log will be the usual /var/log/asterisk/ directory. Thanks, Vivek On 11/30/07, Newbie <[EMAIL PROTECTED]> wrote: Hi, there is no problem with X-Lite, the problem is SPA-3102 shown: Line 1: Registration Status: Failed PSTN Line 1: Registration Status: Failed I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow.. since I am very new with this.. I don't know why this problem occurs could any body please help? Thanks & Regards Winanjaya [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm allow=ulaw allow=h261 allow=h263 allow=h263p register=998:[EMAIL PROTECTED]/998 register=999:[EMAIL PROTECTED]/999 [line1] type=peer host=dynamic defaultip=172.16.1.74 fromuser=998 secret=1234 fromdomain=172.16.1.169 [line2] type=peer host=dynamic defaultip=172.16.1.74 username=999 secret=1234 fromdomain=172.16.1.169 Command> sip show peers Name/username HostDyn Nat ACL Port Status pstnline1/999 (Unspecified)D 0Unmonitored line1 (Unspecified)D 0Unmonitored 250/250172.16.1.88 D 27778Unmonitored 2500 (Unspecified)D 0Unmonitored 251(Unspecified)D 0Unmonitored 5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] - Original Message - From: Vivek Shrivastava To: Newbie ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December 01, 2007 11:34 AM Subject: Re: [asterisk-users] Registration state: Failed Hi, x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues. Thanks, Vivek On 11/30/07, Newbie <[EMAIL PROTECTED] > wrote: Dear Support, I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line. I have 3 extensions: 250 -> my extension 998 -> I configured as Line 1 in SPA-3102 999 -> I configured as PSTN Line 1 in SPA-3102 I have created 998 and 999 to the user extension list of the AsteriskNow why I still got Registration state: Failed for both Line 1 status and PSTN Line status ? my topology is below: Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line Please help Thanks a lot in advance Regards Winanjaya ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.
Re: [asterisk-users] Shared line appearance phones?
On Nov 29, 2007 5:49 AM, Mark Wiater <[EMAIL PROTECTED]> wrote: > Russell Bryant wrote: > > Ron McCarthy wrote: > >> Asterisk 1.4 im guessing? I did not know the Snom's worked with that, > >> Ill have to check it out then! > > > > The way it is implemented in Asterisk is a bit interesting. It uses the > > existing device state support (hints, BLF) to manage the buttons for > shared > > lines. Asterisk changes the state of these virtual "shared lines" to > different > > states, and the light on the phone reflects the state (in use, ringing, > on hold). > > I fought with this in 1.4.5 with polycom phones. I was hoping to share a > DID from a PRI on several > Polycom IP430's. > > Might you be willing to share some specific configurations for such a > situation? > > Mark, That's what I have been trying very unsuccessfully to do as well. It seems to be something that can't be done in a few minutes here and there of spare time :-) > thanks > > mark > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Lacy Moore Somewhere I wish I wasn't ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration state: Failed
well, then i would recommend to see "full" log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with "full=>" in the logger.conf... the log will be the usual /var/log/asterisk/ directory. Thanks, Vivek On 11/30/07, Newbie <[EMAIL PROTECTED]> wrote: > > Hi, > there is no problem with X-Lite, the problem is SPA-3102 shown: > > Line 1: > Registration Status: Failed > > PSTN Line 1: > Registration Status: Failed > > I also had added 1 more extension 251..then tried to call 251 from 250 by > using X-Lite and it works perfectly.. so that's why I am sure there is no > problem with X-Lite .. what I suspect is the problem on Registration process > in AsteriskNow.. > > since I am very new with this.. I don't know why this problem occurs ... > could any body please help? > > Thanks & Regards > Winanjaya > > [general] > context=default > allowoverlap=no > bindport=5060 > bindaddr=0.0.0.0 > srvlookup=yes > videosupport=yes > disallow=all > allow=ilbc > allow=gsm > allow=ulaw > allow=h261 > allow=h263 > allow=h263p > register=998:[EMAIL PROTECTED]/998 > register=999:[EMAIL PROTECTED]/999 > [line1] > type=peer > host=dynamic > defaultip=172.16.1.74 > fromuser=998 > secret=1234 > fromdomain=172.16.1.169 > > [line2] > type=peer > host=dynamic > defaultip=172.16.1.74 > username=999 > secret=1234 > fromdomain=172.16.1.169 > > > Command>* sip show peers* > > Name/username HostDyn Nat ACL Port Status > pstnline1/999 (Unspecified)D 0Unmonitored > line1 (Unspecified)D 0Unmonitored > 250/250172.16.1.88 D 27778Unmonitored > 2500 (Unspecified)D 0Unmonitored > 251(Unspecified)D 0Unmonitored > 5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] > > > > > > > > > - Original Message - > > *From:* Vivek Shrivastava <[EMAIL PROTECTED]> > *To:* Newbie <[EMAIL PROTECTED]> ; Asterisk Users Mailing List - > Non-Commercial Discussion > *Sent:* Saturday, December 01, 2007 11:34 AM > *Subject:* Re: [asterisk-users] Registration state: Failed > > > Hi, > > x-lite has extensive debug facility you can turn that on in the advanced > options, that probably will give better understanding as what is going on > from x-lite side. i also have experienced the same but that involved > firewall and NAT issues. > > Thanks, > > Vivek > > > On 11/30/07, Newbie <[EMAIL PROTECTED]> wrote: > > > > Dear Support, > > > > I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected > > with PSTN line. > > > > I have 3 extensions: > > > > 250 -> my extension > > 998 -> I configured as Line 1 in SPA-3102 > > 999 -> I configured as PSTN Line 1 in SPA-3102 > > > > I have created 998 and 999 to the user extension list of the AsteriskNow > > > > why I still got Registration state: Failed for both Line 1 status and > > PSTN Line status ? > > > > > > my topology is below: > > > > Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line > > > > Please help > > > > Thanks a lot in advance > > > > Regards > > Winanjaya > > > > > > > > ___ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration state: Failed
Hi, there is no problem with X-Lite, the problem is SPA-3102 shown: Line 1: Registration Status: Failed PSTN Line 1: Registration Status: Failed I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow.. since I am very new with this.. I don't know why this problem occurs ... could any body please help? Thanks & Regards Winanjaya [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm allow=ulaw allow=h261 allow=h263 allow=h263p register=998:[EMAIL PROTECTED]/998 register=999:[EMAIL PROTECTED]/999 [line1] type=peer host=dynamic defaultip=172.16.1.74 fromuser=998 secret=1234 fromdomain=172.16.1.169 [line2] type=peer host=dynamic defaultip=172.16.1.74 username=999 secret=1234 fromdomain=172.16.1.169 Command> sip show peers Name/username HostDyn Nat ACL Port Status pstnline1/999 (Unspecified)D 0Unmonitored line1 (Unspecified)D 0Unmonitored 250/250172.16.1.88 D 27778Unmonitored 2500 (Unspecified)D 0Unmonitored 251(Unspecified)D 0Unmonitored 5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] - Original Message - From: Vivek Shrivastava To: Newbie ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, December 01, 2007 11:34 AM Subject: Re: [asterisk-users] Registration state: Failed Hi, x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues. Thanks, Vivek On 11/30/07, Newbie <[EMAIL PROTECTED]> wrote: Dear Support, I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line. I have 3 extensions: 250 -> my extension 998 -> I configured as Line 1 in SPA-3102 999 -> I configured as PSTN Line 1 in SPA-3102 I have created 998 and 999 to the user extension list of the AsteriskNow why I still got Registration state: Failed for both Line 1 status and PSTN Line status ? my topology is below: Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line Please help Thanks a lot in advance Regards Winanjaya ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration state: Failed
Hi, x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues. Thanks, Vivek On 11/30/07, Newbie <[EMAIL PROTECTED]> wrote: > > Dear Support, > > I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected > with PSTN line. > > I have 3 extensions: > > 250 -> my extension > 998 -> I configured as Line 1 in SPA-3102 > 999 -> I configured as PSTN Line 1 in SPA-3102 > > I have created 998 and 999 to the user extension list of the AsteriskNow > > why I still got Registration state: Failed for both Line 1 status and PSTN > Line status ? > > > my topology is below: > > Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line > > Please help > > Thanks a lot in advance > > Regards > Winanjaya > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using existing extensions.conf macros, and co-habitation
Tilghman Lesher wrote: > On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote: > >> [snip] >> The issue is that I have, per "virtual pbx" (i.e. home or business), two >> contexts that these get used from. The "internal-xyzzy" and >> "incoming-xyzzy" contexts (one for each pbx, ie. "xyzzy" is "home" or else >> it's "office"). >> >> I was wondering if there wasn't a more flexible solution to this issue, >> than hard-coding a "Goto(default,s,1)" into them (I have no default >> context, because it would be meaningless). >> >> Perhaps using "Gosub" and "Return". Or do I need to hack the macro, and >> pass in a 3rd argument (bletch)? >> > > MacroExit or Gosub/Return would certainly be possibilities. > > The main thing to note is that this macro that you call standard is actually > just an arbitrary example. It is by no means perfect, so feel free to adapt > it to your own liking. > Sure. I just figured that it would be nice if the canned macros worked out-of-the-box without modification, in the real world. I suppose I could file a bug, and then submit patches for the macro and documentation... -Philip ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using existing extensions.conf macros, and co-habitation
On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote:
> I'm trying to set up my extensions.conf file using some of the existing
> macros like stdexten, etc. while at the same time having two logically
> separate virtual PBX's (with no "default" context) and two trunks coming
> into separate contexts, i.e. one for residence and one for my at-home
> business.
>
> I noticed, however, that macro-stdexten depends on the "default" context:
>
> [macro-stdexten];
> ;
> ; Standard extension macro:
> ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well)
> ; ${ARG2} - Device(s) to ring
> ;
> exten => s,1,Dial(${ARG2},20) ; Ring the
> interface, 20 seconds maximum
> exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
> status
> (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
>
> exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send
> to
> voicemail w/ unavail announce exten => s-NOANSWER,2,Goto(default,s,1)
> ;
> If they press #, return to start
>
> exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
> voicemail w/
> busy announce exten => s-BUSY,2,Goto(default,s,1) ; If
> they press #,
> return to start
>
> exten => _s-.,1,Goto(s-NOANSWER,1); Treat
> anything else as no answer
>
> exten => a,1,VoicemailMain(${ARG1})
>
>
> The issue is that I have, per "virtual pbx" (i.e. home or business), two
> contexts that these get used from. The "internal-xyzzy" and
> "incoming-xyzzy" contexts (one for each pbx, ie. "xyzzy" is "home" or else
> it's "office").
>
> I was wondering if there wasn't a more flexible solution to this issue,
> than hard-coding a "Goto(default,s,1)" into them (I have no default
> context, because it would be meaningless).
>
> Perhaps using "Gosub" and "Return". Or do I need to hack the macro, and
> pass in a 3rd argument (bletch)?
MacroExit or Gosub/Return would certainly be possibilities.
The main thing to note is that this macro that you call standard is actually
just an arbitrary example. It is by no means perfect, so feel free to adapt
it to your own liking.
--
Tilghman
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Re: [asterisk-users] Copy or Make + Make Install
On Friday 30 November 2007 17:33:09 Mojo with Horan & Company, LLC wrote: > Tzafrir Cohen wrote: > > On Wed, Nov 28, 2007 at 10:47:44AM -0900, Mojo with Horan & Company, LLC wrote: > >> You might want the directory structure at /var/lib/asterisk as well, as > >> it contains the current state of the voicemail boxes and any custom > >> sound files that might have been added > > > > Voicemail boxes are actually under /var/spool/voicemail . > > Ah, yes, of course. Thank you :) Or actually under /var/spool/asterisk/voicemail -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Registration state: Failed
Dear Support, I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line. I have 3 extensions: 250 -> my extension 998 -> I configured as Line 1 in SPA-3102 999 -> I configured as PSTN Line 1 in SPA-3102 I have created 998 and 999 to the user extension list of the AsteriskNow why I still got Registration state: Failed for both Line 1 status and PSTN Line status ? my topology is below: Users <--> AsteriskNow <--> SPA-3102 <--> PSTN line Please help Thanks a lot in advance Regards Winanjaya ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] To DB or not to DB?
> "PK" == Philipp Kempgen <[EMAIL PROTECTED]> writes: PK> Anthony Francis wrote: >> 2. Many features such as hinting (BLF) do not work with >> realtime. PK> That's only true if *extensions.conf* comes from a db table. Nope, turn off caching and use realtime for SIP peers, and suddenly BLF doesn't work. At least that is how it is in 1.2.x. The caching negates some of the advantages of using realtime. /Benny ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using existing extensions.conf macros, and co-habitation
Brian J. Murrell wrote: > On Fri, 2007-11-30 at 15:08 -0800, Philip Prindeville wrote: > >> bump... >> > > What's with all this "bump" I see here? Is this a web forum? > > b. > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Somebody asked a question and no one answered. A bump is just a nudge to politely ask this is the 2nd time I have asked this does someone know the answer. I have used the before and it usually works. Lyle ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using existing extensions.conf macros, and co-habitation
On Fri, 2007-11-30 at 15:08 -0800, Philip Prindeville wrote: > bump... What's with all this "bump" I see here? Is this a web forum? b. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality
Thank you much for the prompt reply Salvatore. Would you have the time to explain further how should I go for verifying that SDP and RTP are OK. Also what is reffered to as the TDM site. Veselin On Fri, Nov 30, 2007 at 05:01:17PM -0500, Salvatore Giudice wrote: > Take a packet capture of your VoIP segment and verify that the SDP is > correct and that the RTP is making it to the correct places. If all that > looks good and this is a straight out quality problem, then you need to > figure out if it's happening on the voip side or on the TDM side. You should > make calls (with captures) VoIP to Voip passing the media through your > asterisk and also try routing a tdm call in and back out. If you have the > equipment, take a mos score of the TDM loop. > > Without any of the above, you will not be able to isolate the issue. > > -- > Salvatore Giudice > [EMAIL PROTECTED] > > VoIP Security Training, LLC > http://VoIPSecurityTraining.com > > 848 N. Rainbow Blvd. #1676 > Las Vegas, NV 89107 > Phone: (617) 959-7625 > Fax: (214) 279-2906 > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Veselin > Kantsev > Sent: Friday, November 30, 2007 2:47 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality > > Hello, > I have an Asterisk running with a Sangoma A200 card with Hardware Echo > cancelling connected to the UK PSTN. > If a PSTN call comes in, voice both ways is OK, however if an outgoing > call over the PSTN is made I can hear the other party OK but they can > not, they can barely understand what I am saying, my voice is unclear > fading and skipping. > Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 > are OK too. I've tried gsm/ulaw/alaw codecs so far. > Tried disabling the echo cancelling as well. > > Any suggestions will be greatly appreciated. > > > Regards, > Veselin > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as non-root/best practices
Well, there you go then - either add /usr/sbin to your path, or provide a full path thusly: /usr/sbin/asterisk -r CP Robert McNaught wrote: > not in path > > [EMAIL PROTECTED] echo $PATH > /usr/kerberos/bin:/usr/lib/courier-imap/bin:/usr/local/bin:/bin:/usr/bin:/usr/X11R6/bin:/home/admin/bin > >> >> Is /sbin in your path? >> >> CP >> >> Robert McNaught wrote: >> > >> > my problem is that a non-privileged user, eg admin, cannot log in and >> > connect to the console by issuing the following >> > >> > [EMAIL PROTECTED] asterisk -r >> > bash: asterisk: command not found >> > >> > [EMAIL PROTECTED] whereis asterisk >> > asterisk: /usr/sbin/asterisk /usr/lib/asterisk /usr/include/asterisk >> > /usr/include/asterisk.h /usr/share/man/man8/asterisk.8 >> > >> > what is the best way to solve this problem? >> > >> > i have tried adding >> > >> > admin ALL=(ALL) ALL- I will prune back once I verify I can >> > get this working >> > >> > into visudo, but even that returns asterisk:command not found >> > >> > Does anyone out there know the best way around this - I tried adding in >> > a symbolic link in /usr/bin/asterisk to point to the >> > /home/asterisk/asterisk-bin/sbin/asterisk file, which worked, but is a >> > hack around the problem and don't believe this is the way >> > >> > It seems that non-privileged users cannot run commands in sbin, but can >> > in bin directories >> > >> > Robert ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729/MOH Quality
If the majority of the MoH is queues, move the location of the queue. On Nov 28, 2007 4:42 PM, Darryl Dunkin <[EMAIL PROTECTED]> wrote: > Does anyone have any opinions on the music on hold quality over G729? > The stock files seem to sound terrible over it, this is enhanced further > by calls coming from the PSTN via a Zaptel gateway. I am only using the > stock wav files and have not attempted to use much else so far. > > I've ruled out timing issues on the system generating the MOH itself > (ztdummy on the PBX itself, our Zaptel gateway is a separate Asterisk > server). There is no transcoding going on in the middle except via our > Zaptel/T1 gateway. When using G711 it sounds fine of course, but this > doesn't work well for remote sites with lower bandwidth connections. > > As of now, I'm torn between getting complaints from end users about the > music or killing it entirely (leaving people waiting in queues with dead > silence). > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Correct syntax for IF()?
On Fri, 30 Nov 2007 00:30:06 -0500, Jared Smith <[EMAIL PROTECTED]>
wrote:
>Sounds like a perfect application for the ISNULL dialplan function. Of
>course, that adds a whole new set of curly braces and parentheses to
>watch out for.
Thanks Jared for the pointer :-)
exten => s,1,Set(foo=${ISNULL(${var1})})
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+isnull
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Re: [asterisk-users] Shared line appearance phones?
Mark Wiater wrote: > I fought with this in 1.4.5 with polycom phones. I was hoping to share a DID > from a PRI on several > Polycom IP430's. > > Might you be willing to share some specific configurations for such a > situation? There are some basic examples in doc/sla.pdf in the 1.4 tree. However, I have on my to-do list to spend a week with an SLA test environment and coming up with an extensive set of examples of the different ways it can be used. I will post something to this list when that is available. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as non-root/best practices
On Fri, 30 Nov 2007, Robert McNaught wrote: >> It seems that non-privileged users cannot run commands in sbin, but >> can in bin directories Unless something in your host is major league hosed, this is not true. Try: /sbin/runlevel /usr/sbin/ntpdate -q 0.us.pool.ntp.org Depending on who you ask, the "s" in sbin means "static" or "system." On Linux, it appears to mean "system" since both of these are dynamically linked. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SLA: Handling of errors in outgoing call
Steve Langstaff wrote: > [line1_outbound] > exten => disa,1,Disa(no-password|line1_outbound) > exten => _,1,Dial(SIP/[EMAIL PROTECTED]) > exten => _,2,Hangup > So to summarise: > if I seize the line and dial a number known at vsp5000 then I > get ringing etc - good. > if I seize the line and dial a number unknown at vsp5000 then > the call drops silently - not good. Your issue is actually in the [line1_outbound] context that I have quoted above. Here is what happens: As far as Disa() is concerned, any 4 digit number is a valid extension. Once you dial something (whether is valid at the other end or not), the call goes on and executes Dial(). However, since the number you have dialed is not valid, Dial() immediately returns and then Hangup is executed. That is when the call is dropping. So, I can think of a few different ways to solve this issue. The first couple involve using an IAX2 or DUNDi switch statement in the line1_outbound context. That would allow Asterisk to query the remote server as to what extensions are valid. However, I won't get into the details of how that is configured right now ... The other alternative is to solve it in the dialplan, with something like this: [line1_outbound] exten => disa,1,Disa(no-password|line1_outbound) exten => _,1,Dial(SIP/[EMAIL PROTECTED]) exten => _,n,Congestion exten => _,n,Wait(10) ; Give Congestion for 10 seconds exten => _,n,Hangup You could improve this even further by checking the DIALSTATUS and playing different tones, or just hanging up, accordingly. I hope this helps, -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
Griefs? rejected connect attempt from 111.111.111.111, who was trying to reach '12345678' No authority found call rejected by 111.111.111.111: No authority found But once it works it works... I have DTMF issues with sending calls from 1.2 to what I suspect is a really old 1.4 build via IAX that then hands those calls off as SIP . But I suspect it could be fixed in the SIP configuration. This is a very isolated situation. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Only call me once
Store a value indicating it has been called as a unique key in AstDB, and set your dial plan to check for it. On Fri, 30 Nov 2007, [EMAIL PROTECTED] wrote: > Anyone have an idea how to implement a phone number that can only be > called once? The first time it will process normally and any > subsequent calls will be rejected. > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as non-root/best practices
not in path [EMAIL PROTECTED] echo $PATH /usr/kerberos/bin:/usr/lib/courier-imap/bin:/usr/local/bin:/bin:/usr/bin:/usr/X11R6/bin:/home/admin/bin > > Is /sbin in your path? > > CP > > Robert McNaught wrote: > > > > my problem is that a non-privileged user, eg admin, cannot log in and > > connect to the console by issuing the following > > > > [EMAIL PROTECTED] asterisk -r > > bash: asterisk: command not found > > > > [EMAIL PROTECTED] whereis asterisk > > asterisk: /usr/sbin/asterisk /usr/lib/asterisk /usr/include/asterisk > > /usr/include/asterisk.h /usr/share/man/man8/asterisk.8 > > > > what is the best way to solve this problem? > > > > i have tried adding > > > > admin ALL=(ALL) ALL- I will prune back once I verify I can > > get this working > > > > into visudo, but even that returns asterisk:command not found > > > > Does anyone out there know the best way around this - I tried adding in > > a symbolic link in /usr/bin/asterisk to point to the > > /home/asterisk/asterisk-bin/sbin/asterisk file, which worked, but is a > > hack around the problem and don't believe this is the way > > > > It seems that non-privileged users cannot run commands in sbin, but can > > in bin directories > > > > Robert > > > > > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Only call me once
Anyone have an idea how to implement a phone number that can only be called once? The first time it will process normally and any subsequent calls will be rejected. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is it better to use debian binary or compiled version?
Hi. I am starting with asterisk, but I will not have problem to compile the newer version 1.4. My question is if it is worth to compile rather then using the binary 1.2 version in Debian stable? I plan to use one analog PSTN line and two sip providers. Thanks Jiri ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sidetone
Todd wrote: > Hi - > I've got a new install with a Sangoma A200 and a few GXP2000's. When > users are talking over the Sangoma, they get a lot of sidetone (local > echo). Internal calls are fine. Where do I adjust that? I assume > its in zapata.conf somewhere? > thanks > Todd > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > I have had great experience with the oslec echo canceller, although it's more difficult than modifying zapata.conf, it has seemed more effective and worth the install. http://www.rowetel.com/ucasterisk/oslec Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] My AsteriskNo unable to registration
Dear The Expert, I am very new with this, I have installed AsteriskNow, X-Lite as my SoftPhone, I am using SPA-3102. I have 3 extensions, me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below) My problem is, I am unable to call 998, I thought this is registration problem, (because the Linksys screen info said Registration Failed) Could any body please help? Many thanks in advance Regards Bie below is my sip.conf allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm I also had 2 extensions (me at 250 and 998 is my SPA-3102) and my users.conf goes below: [general] fullname=New User userbase=6000 hasvoicemail=yes vmsecret=1234 hassip=yes hasiax=yes hasmanager=no callwaiting=yes threewaycalling=yes callwaitingcallerid=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callgroup=1 pickupgroup=1 host=dynamic localextenlength=0 allow_aliasextns=no allow_an_extns=no hasagent=no hasdirectory=no [250] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=Winanjaya group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=250 secret=1234 threewaycalling=yes vmsecret=1234 zapchan= registeriax=yes registersip=yes canreinvite=no nat=no dtmfmode=rfc2833 disallow=all allow=all type=peer [998] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=MyLine1 group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=999 secret=1234 threewaycalling=yes vmsecret=1234 zapchan= registeriax=yes registersip=yes canreinvite=no nat=no dtmfmode=rfc2833 disallow=all allow=all type=peer [999] callwaiting=yes cid_number= context=numberplan-custom-2 email= fullname=MyPSTN group= hasagent=yes hasdirectory=no hasiax=yes hasmanager=no hassip=yes hasvoicemail=yes host=dynamic mailbox=999 secret=1234 threewaycalling=yes vmsecret=1234 zapchan= registeriax=yes registersip=yes canreinvite=no nat=no dtmfmode=rfc2833 disallow=all allow=all type=peer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem registering Cisco 7970 phone with Asterisk 1.4 running FreePBX
Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install smoothly but I am stuck at the registration part. I went through another post here on this subject at : http://forums.digium.com/viewtopic.php?t=15212&highlight=7970 This helped me get past the SIP 401 Unauthorized error when I went into the sip_additional.conf file and changed the "secret=" line to "password=" However, the phone is still stuck in Registering, and I see these new messages on the asterisk CLI : <-> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.16.121.170 : 49309 (NAT) <--- Transmitting (NAT) to 10.16.121.170:49309 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f;received=10.16.121.170 From: ;tag=001e4a5f1272ab51cff4-e26d9841 To: Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <> d2armyFreePBX*CLI> <--- Transmitting (NAT) to 10.16.121.170:49309 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f;received=10.16.121.170 From: ;tag=001e4a5f1272ab51cff4-e26d9841 To: ;tag=as3f746d9f Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: ;expires=3600 Date: Thu, 29 Nov 2007 14:00:55 GMT Content-Length: 0 <> Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) Retransmitting #1 (NAT) to 10.16.121.170:49309: OPTIONS sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.19.125.13:5060;branch=z9hG4bK1d9fde6c;rport From: "Unknown" ;tag=as76e8e4a2 To: Contact: Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 29 Nov 2007 14:00:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- d2armyFreePBX*CLI> <--- SIP read from 10.16.121.170:49309 ---> REGISTER sip:172.19.125.13 SIP/2.0 Via: SIP/2.0/UDP 10.16.121.170:5060;branch=z9hG4bK12e80d0f From: ;tag=001e4a5f1272ab51cff4-e26d9841 To: Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 Date: Fri, 02 Nov 2007 23:25:54 GMT CSeq: 101 REGISTER User-Agent: Cisco-CP7970G/8.3.0 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="30006" Supported: (null),X-cisco-xsi-6.0.2 Content-Length: 0 Expires: 3600 These messages repeat again and again. It does not look like the "SIP/2.0 200 OK" message is any better than 401 before. My config in sip_additional.conf is : [2001] type=friend password=2001 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 pickupgroup= nat=yes [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow= dial=SIP/2001 context=from-sip canreinvite=no callgroup= callerid=device <2001> allow= accountcode= call-limit=50 My updated SEP file for this hard phone is at http://cid-ff3ef0764138e401.skydrive.live.com/self.aspx/Public/SEP001E4A5F1270.cnf.xml On the phone side when I ssh in, "show register" shows : LINE REGISTRATION TABLE Proxy Registration: ENABLED, state: IDLE line APR state timer expires proxy:port --- - -- -- 1 .1x REGISTERING 0 0 172.19.125.13:5060 2 ... NONE 0 0 undefined:0 3 ... NONE 0 0 undefined:0 4 ... NONE 0 0 undefined:0 5 ... NONE 0 0 undefined:0 6 ... NONE 0 0 undefined:0 7 ... NONE 0 0 undefined:0 8 ... NONE 0 0 undefined:0 1-BU .1x REGISTERING 3600 17 172.19.125.13:5060 Note: APR is Authenticated, Provisioned, Registered Please help, thanks John _ You keep typing, we keep giving. Download Messenger and join the i’m Initiative now. http://im.live.com/messenger/im/home/?source=TAGLM___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Copy or Make + Make Install
Tzafrir Cohen wrote: > On Wed, Nov 28, 2007 at 10:47:44AM -0900, Mojo with Horan & Company, LLC > wrote: > >> You might want the directory structure at /var/lib/asterisk as well, as >> it contains the current state of the voicemail boxes and any custom >> sound files that might have been added >> > > Voicemail boxes are actually under /var/spool/voicemail . > > Ah, yes, of course. Thank you :) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] v33 of codec_g729a released
Version 33 of codec_g729a for Asterisk 1.4 has been released. This release is a compatibility update to work with the latest version of Asterisk. Users of this module upgrading to Asterisk 1.4.15 will need to upgrade to this version of codec_g729a. The module is available for download at the following location: http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/ Thank you! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as non-root/best practices
Robert McNaught wrote:
>> thanks for the reply Tzafrir,
>>
>> I tried the below, but I think maybe I misexplained what I am trying
>> to do. I have asterisk running as user asterisk - I followed the
>> instructions in the Asterisk book and have everything stored in
>> /home/asterisk/asterisk-bin - this includes logs, pid files, configs
>> etc etc
I can't see why you are still having trouble. There are lots of
explanations for how to do this.
Why are you trying to build asterisk under /home?
The binaries should be in your usual path (/bin, /sbin, /usr/bin), the
configuration stuff should *always* be in /etc/asterisk, and build
asterisk to store all voice mail and logs under /var/{lib,log,spool}.
I have documented this process here:
http://www.theopensourcerer.com/2007/10/30/untangle-asterisk-pbx-and-file-server-all-in-one-part-7/
I don't mean to be rude, but if you really can't do this - pay someone
who can...
Al
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[asterisk-users] Asterisk-addons 1.4.5 Released
The Asterisk.org development team has released Asterisk-addons version 1.4.5. This release contains a few bug fixes, but is required for compatibility with the latest version of Asterisk, 1.4.15. Thank you for your support! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using existing extensions.conf macros, and co-habitation
bump...
Philip Prindeville wrote:
> I'm trying to set up my extensions.conf file using some of the existing
> macros like stdexten, etc. while at the same time having two logically
> separate virtual PBX's (with no "default" context) and two trunks coming
> into separate contexts, i.e. one for residence and one for my at-home
> business.
>
> I noticed, however, that macro-stdexten depends on the "default" context:
>
> [macro-stdexten];
> ;
> ; Standard extension macro:
> ; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well)
> ; ${ARG2} - Device(s) to ring
> ;
> exten => s,1,Dial(${ARG2},20) ; Ring the
> interface, 20 seconds maximum
> exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
> status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
>
> exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send
> to voicemail w/ unavail announce
> exten => s-NOANSWER,2,Goto(default,s,1) ; If they press
> #, return to start
>
> exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
> voicemail w/ busy announce
> exten => s-BUSY,2,Goto(default,s,1) ; If they press #,
> return to start
>
> exten => _s-.,1,Goto(s-NOANSWER,1); Treat
> anything else as no answer
>
> exten => a,1,VoicemailMain(${ARG1})
>
>
> The issue is that I have, per "virtual pbx" (i.e. home or business), two
> contexts
> that these get used from. The "internal-xyzzy" and "incoming-xyzzy" contexts
> (one
> for each pbx, ie. "xyzzy" is "home" or else it's "office").
>
> I was wondering if there wasn't a more flexible solution to this issue, than
> hard-coding a "Goto(default,s,1)" into them (I have no default context,
> because it
> would be meaningless).
>
> Perhaps using "Gosub" and "Return". Or do I need to hack the macro, and pass
> in a
> 3rd argument (bletch)?
>
> Is this doable?
>
> Thanks,
>
> -Philip
>
>
>
>
>
>
>
>
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Re: [asterisk-users] asterisk as non-root/best practices
Is /sbin in your path? CP Robert McNaught wrote: > > my problem is that a non-privileged user, eg admin, cannot log in and > connect to the console by issuing the following > > [EMAIL PROTECTED] asterisk -r > bash: asterisk: command not found > > [EMAIL PROTECTED] whereis asterisk > asterisk: /usr/sbin/asterisk /usr/lib/asterisk /usr/include/asterisk > /usr/include/asterisk.h /usr/share/man/man8/asterisk.8 > > what is the best way to solve this problem? > > i have tried adding > > admin ALL=(ALL) ALL- I will prune back once I verify I can > get this working > > into visudo, but even that returns asterisk:command not found > > Does anyone out there know the best way around this - I tried adding in > a symbolic link in /usr/bin/asterisk to point to the > /home/asterisk/asterisk-bin/sbin/asterisk file, which worked, but is a > hack around the problem and don't believe this is the way > > It seems that non-privileged users cannot run commands in sbin, but can > in bin directories > > Robert > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as non-root/best practices
> thanks for the reply Tzafrir, > > I tried the below, but I think maybe I misexplained what I am trying > to do. I have asterisk running as user asterisk - I followed the > instructions in the Asterisk book and have everything stored > in /home/asterisk/asterisk-bin - this includes logs, pid files, > configs etc etc > > my asterisk.conf is > > [directories] > astetcdir => /home/asterisk/asterisk-bin/asterisk > astmoddir => /home/asterisk/asterisk-bin/lib/asterisk/modules > astvarlibdir => /home/asterisk/asterisk-bin/lib/asterisk > astdatadir => /home/asterisk/asterisk-bin/lib/asterisk > astagidir => /home/asterisk/asterisk-bin/lib/asterisk/agi-bin > astspooldir => /home/asterisk/asterisk-bin/spool/asterisk > astrundir => /home/asterisk/asterisk-bin/run > astlogdir => /home/asterisk/asterisk-bin/log/asterisk > > [options] > ;internal_timing = yes > systemname = X ; prefix uniqueid with a system name for global > uniqueness issues > ; Changing the following lines may compromise your security. > ;[files] > ;astctlpermissions = 0770 > astctlowner = asterisk > astctlgroup = asterisk > ;astctl = asterisk.ctl > > my problem is that a non-privileged user, eg admin, cannot log in and > connect to the console by issuing the following > > [EMAIL PROTECTED] asterisk -r > bash: asterisk: command not found > > [EMAIL PROTECTED] whereis asterisk > asterisk: /usr/sbin/asterisk /usr/lib/asterisk /usr/include/asterisk > /usr/include/asterisk.h /usr/share/man/man8/asterisk.8 > > what is the best way to solve this problem? > > i have tried adding > > admin ALL=(ALL) ALL- I will prune back once I verify I can > get this working > > into visudo, but even that returns asterisk:command not found > > Does anyone out there know the best way around this - I tried adding > in a symbolic link in /usr/bin/asterisk to point to > the /home/asterisk/asterisk-bin/sbin/asterisk file, which worked, but > is a hack around the problem and don't believe this is the way > > It seems that non-privileged users cannot run commands in sbin, but > can in bin directories > > Robert > > > > > > On Mon, Nov 19, 2007 at 08:51:21AM -0800, Robert McNaught wrote: > > > Hi, > > > > > > I have set up asterisk to run as non root, and allow admin users to log > > > in to the server as asterisk, which gives them privileges to edit > > > configs in the asterisk home directory. > > > > The daemon runs as the user asterisk. There is no reason why the admin > > should run as the user asterisk. > > > > > > > > As for connecting to the console with 'asterisk -r' - this by default > > > does not work as asterisk is owned stored in /usr/sbin/asterisk > > > > > > I am reading that the best way to solve this is to use 'visudo' - I > > > added this:- > > > > > > asteriskALL=/usr/sbin/asterisk -r NOPASSWD: ALL > > > > > > This is totally unrequired. You just need to set proper permissions for > > the socket /var/run/asterisk/asterisk.ctl . This is done in > > asterisk.conf - > > > > [files] > > ;astctlpermissions = 0660 > > ;astctlowner = root > > astctlgroup = asterisk > > ;astctl = asterisk.ctl > > > > http://svn.digium.com/svn/asterisk/branches/1.4/doc/asterisk-conf.txt > > > > > asteriskALL=/usr/sbin/safe_asterisk NOPASSWD: ALL > > > > Why would Asterisk need to run safe_asterisk? > > > > With an arbitrary parameter? > > > > You may want to permit some administrator to do that, but not the > > asterisk daemon. This probably opens the door to priviliges escalations. > > > > -- > >Tzafrir Cohen > > icq#16849755 jabber:[EMAIL PROTECTED] > > +972-50-7952406 mailto:[EMAIL PROTECTED] > > http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir > > > > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk as non-root/best practices
thanks for the reply Tzafrir, I tried the below, but I think maybe I misexplained what I am trying to do. I have asterisk running as user asterisk - I followed the instructions in the Asterisk book and have everything stored in /home/asterisk/asterisk-bin - this includes logs, pid files, configs etc etc my asterisk.conf is [directories] astetcdir => /home/asterisk/asterisk-bin/asterisk astmoddir => /home/asterisk/asterisk-bin/lib/asterisk/modules astvarlibdir => /home/asterisk/asterisk-bin/lib/asterisk astdatadir => /home/asterisk/asterisk-bin/lib/asterisk astagidir => /home/asterisk/asterisk-bin/lib/asterisk/agi-bin astspooldir => /home/asterisk/asterisk-bin/spool/asterisk astrundir => /home/asterisk/asterisk-bin/run astlogdir => /home/asterisk/asterisk-bin/log/asterisk [options] ;internal_timing = yes systemname = X ; prefix uniqueid with a system name for global uniqueness issues ; Changing the following lines may compromise your security. ;[files] ;astctlpermissions = 0770 astctlowner = asterisk astctlgroup = asterisk ;astctl = asterisk.ctl my problem is that a non-privileged user, eg admin, cannot log in and connect to the console by issuing the following [EMAIL PROTECTED] asterisk -r bash: asterisk: command not found [EMAIL PROTECTED] whereis asterisk asterisk: /usr/sbin/asterisk /usr/lib/asterisk /usr/include/asterisk /usr/include/asterisk.h /usr/share/man/man8/asterisk.8 what is the best way to solve this problem? i have tried adding admin ALL=(ALL) ALL- I will prune back once I verify I can get this working into visudo, but even that returns asterisk:command not found Does anyone out there know the best way around this - I tried adding in a symbolic link in /usr/bin/asterisk to point to the /home/asterisk/asterisk-bin/sbin/asterisk file, which worked, but is a hack around the problem and don't believe this is the way It seems that non-privileged users cannot run commands in sbin, but can in bin directories Robert > > On Mon, Nov 19, 2007 at 08:51:21AM -0800, Robert McNaught wrote: > > Hi, > > > > I have set up asterisk to run as non root, and allow admin users to log > > in to the server as asterisk, which gives them privileges to edit > > configs in the asterisk home directory. > > The daemon runs as the user asterisk. There is no reason why the admin > should run as the user asterisk. > > > > > As for connecting to the console with 'asterisk -r' - this by default > > does not work as asterisk is owned stored in /usr/sbin/asterisk > > > > I am reading that the best way to solve this is to use 'visudo' - I > > added this:- > > > > asteriskALL=/usr/sbin/asterisk -r NOPASSWD: ALL > > > This is totally unrequired. You just need to set proper permissions for > the socket /var/run/asterisk/asterisk.ctl . This is done in > asterisk.conf - > > [files] > ;astctlpermissions = 0660 > ;astctlowner = root > astctlgroup = asterisk > ;astctl = asterisk.ctl > > http://svn.digium.com/svn/asterisk/branches/1.4/doc/asterisk-conf.txt > > > asteriskALL=/usr/sbin/safe_asterisk NOPASSWD: ALL > > Why would Asterisk need to run safe_asterisk? > > With an arbitrary parameter? > > You may want to permit some administrator to do that, but not the > asterisk daemon. This probably opens the door to priviliges escalations. > > -- >Tzafrir Cohen > icq#16849755 jabber:[EMAIL PROTECTED] > +972-50-7952406 mailto:[EMAIL PROTECTED] > http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir > > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outgoing PSTN calls , unusable voice quality
Take a packet capture of your VoIP segment and verify that the SDP is correct and that the RTP is making it to the correct places. If all that looks good and this is a straight out quality problem, then you need to figure out if it's happening on the voip side or on the TDM side. You should make calls (with captures) VoIP to Voip passing the media through your asterisk and also try routing a tdm call in and back out. If you have the equipment, take a mos score of the TDM loop. Without any of the above, you will not be able to isolate the issue. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Veselin Kantsev Sent: Friday, November 30, 2007 2:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Outgoing PSTN calls , unusable voice quality Hello, I have an Asterisk running with a Sangoma A200 card with Hardware Echo cancelling connected to the UK PSTN. If a PSTN call comes in, voice both ways is OK, however if an outgoing call over the PSTN is made I can hear the other party OK but they can not, they can barely understand what I am saying, my voice is unclear fading and skipping. Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 are OK too. I've tried gsm/ulaw/alaw codecs so far. Tried disabling the echo cancelling as well. Any suggestions will be greatly appreciated. Regards, Veselin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suppressing certain queue announcement voiceprompts
What if you set queue-thankyou to empty? queue-thankyou = "" I have a faint memory of doing this in the old 1.0 days... Not sure if it works in the current releases... // T > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: den 30 november 2007 18:08 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Suppressing certain queue > announcement voiceprompts > > > > > > Short of replacing a sound file with a sound file containing > > > > > only a short period of silence, is there any way to suppress > > > > > certain sounds from playing during queue processing by > > > > > configuring for example queues.conf or other similar files? > > > > > > > > Which announcements are you trying to not play? > > > > > > queue-thankyou for instance, to name one. Or any other of the > > > queue-* files in general. From time to time it can be > convenient to > > > change > > the > > > exact prompts played (order and contents) due to language > > > differences and personal preference of the end-users. > > > > The question is more like what exactly do you mean with > "from time to > > time"? > > > > Anyway, your best option is probably to create one or more prompt > > languages by copying the English prompts to a new directory like > > "en2", "en3" and then use Set(LANGUAGE=en3) in the dialplan > when you > > think this is appropriate. For each of these artificial > languages you > > can now decide how to modify the sound files. > > > > Cheers, Philipp > > Again, very good advice thank you Philipp. And probably a > very reasonable way to do this if dynamic behaviour is > needed. But in my case time-to-time was meant as "every once > in a while there is a particullar installation that requires > this". So statically doing this is ok in my case. > > I'll continue with my replace-with-silence-file method for > now. Thanks for the input. > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do While loop
here's a do-while loop - the contents of the loop are executed BEFORE
the condition is tested.
--
labelA:
do some loopy things
if (we need to loop again)
goto labelA:
--
to contrast that, here's a while loop:
--
labelA:
if (we need to do the loop at all)
{
do some loopy things
goto labelA:
}
--
Sorry it's in some pseudocode that doesn't really represent a language
at all. I can't produce asterisk's dialplan functions from memory yet!
I'm sure that this will convert very simply though with minor work. "if"
would become "GotoIf".
For the archives, the usage of asterisk's While construct is found at
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+While
Mojo
Mike wrote:
> Hi,
>
> Is there a way to have a Do-While sort of loop, as opposed to a simple
> While?
>
> I have a condition that the loop depends on even for the first
> iteration, as it often happens in life.
>
> Regards,
>
> Mike
>
>
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[asterisk-users] Outgoing PSTN calls , unusable voice quality
Hello, I have an Asterisk running with a Sangoma A200 card with Hardware Echo cancelling connected to the UK PSTN. If a PSTN call comes in, voice both ways is OK, however if an outgoing call over the PSTN is made I can hear the other party OK but they can not, they can barely understand what I am saying, my voice is unclear fading and skipping. Internal SIP and IAX2 calls are OK, incoming/outgoing calls over IAX2 are OK too. I've tried gsm/ulaw/alaw codecs so far. Tried disabling the echo cancelling as well. Any suggestions will be greatly appreciated. Regards, Veselin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
If you desire SIP in Avaya, you have to add a SES (SIP Enablement Server) to your Avaya setup. They are cheap. You only have to pay for the box and the maintenance percentage. You don't need to buy user ports or any of that garbage as long as you setup your extensions using Optum, which is a free Avaya feature. The SES maintains a registry and a dial plan. SIP phones attached to SES send media directly to medpros and the SES does a protocol conversion between SIP and H.323 to bridge a connection between the SIP phone and the CLAN cards. The voicemail issue you describe with the MWI is because Avaya's systems use qsig trunks to connect to voicemail servers. Asterisk is not connected int hat manner, so of course you won't be able to support Avaya MWI's. However, you can deposit a script on your asterisk that would send the standard notifies to the Avaya phones to manipulate the MWI's directly. However, you will need to statically address the phones and keep track of them because you cannot poll an SES server for their SIP URI's. Hell, Avaya won't even give you root on any of their servers. You cant audit the box and you can't poll them unless you pay them money to join their partner program and get their SDK. If you already have Avaya, you should just buy Message Networking or a Mitel voicemail server if you want seamless voicemail with Avaya. However, you should know that using Avaya is probably a bad idea to begin with. Until February 07, the majority Avaya's soft switch products were running on Redhat 9, which was unsupported since 2003. Avaya was only managing a dozen packages and they've always left it up to the customer to know when they need an update, requiring the customer to request a field load. It has to be the worst update model in the industry when it comes to infrastructure monitoring and patching. By using Avaya, you are blindly trusting them to properly maintain a Linux appliance. This is something they are not capable of and you can't even audit them. Avaya is what happens to organizations when they have ignorant telecom infrastructure engineers deciding what products to buy. Avaya focuses sales on those engineers because they k now their products won't pass certification by network, systems, or security engineers. Telecom engineers only look for features and usually get their asses handed to them after they put Avaya VoIP into their infrastructure. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser Sent: Friday, November 30, 2007 9:54 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Off-Topic: Avaya This is both a hardware and software licensing issue. Avaya offers a SIP server separate from their main VoIP gateway. The core platform uses H.323. Either SIP or H.323 has a license cost per registered device. We have an Avaya S8300 Communications Manager providing H.323 and have this tied to an Asterisk deployment on a Sun Microsystems server. The connection between the two systems are handled by both T1, (PRI using Qsig), and H.323. The BIG issue we have is we cannot light the message waiting light on the Avaya 46XX phones registered to the Avaya server but using Asterisk voice mail. If anyone can help we would pay to solve this. Our Asterisk is 1.2.xx. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Cabrera Obed Sent: Friday, November 30, 2007 7:30 AM To: Asterisk Users Mailing List Subject: [asterisk-users] Off-Topic: Avaya Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - How to add a new TAPI driver on an XP system ?
Hi, To make a long story short, I can't install any TAPI driver on my XP platform. A. Within Config Panel|Modems and Telephony options|Advanced parameters, I've got a list of 7 TAPI drivers. Among them is Omniis TAPI driver for Asterisk. B. I can properly configure this driver (line, context, ...). C. When I open Outlook 2002 Contacts panel, I can select "Call this contact" from Actions menu. D. When the "New call" popup appears, I can click on "Dialing Options ..." E. When the "Dialing options" popup appears, there is a scrolling list "Dialing using line" in which I can find a list of modem drivers but not a single TAPI driver. F. If I check running Services (Config Panel|Administration Tools|Services], Telephony service is said to be running. My questions are: 1. Is there a way to set a TSP driver to be default driver to be used and skip "Dialing options" windows ? 2. Should I see TAPI drivers within "Dialing using line" scrolling list ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
I manage a large Avaya implementation with three systems at different locations. I hate Avaya's manageability, lack of features, and extremely high cost. That's why I'm looking into alternatives to replace the whole thing in a year or two. I would appreciate any other opinions and findings regarding the integration with Avaya and switching from Avaya. Our IP phones are 4600 series as well. Also, I don't think SIP was even supported until CM v3.x, so you're SOL with anything earlier. Jim Houser wrote: > This is both a hardware and software licensing issue. > Avaya offers a SIP server separate from their main VoIP gateway. > The core platform uses H.323. > Either SIP or H.323 has a license cost per registered device. > We have an Avaya S8300 Communications Manager providing H.323 and have this > tied to an Asterisk deployment on a Sun Microsystems server. The connection > between the two systems are handled by both T1, (PRI using Qsig), and H.323. > > The BIG issue we have is we cannot light the message waiting light on the > Avaya 46XX phones registered to the Avaya server but using Asterisk voice > mail. > > If anyone can help we would pay to solve this. Our Asterisk is 1.2.xx. > > Thanks. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro > Cabrera Obed > Sent: Friday, November 30, 2007 7:30 AM > To: Asterisk Users Mailing List > Subject: [asterisk-users] Off-Topic: Avaya > > Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP > or H.323 ??? > > Anybody can't tell me this...so I'm here for thei reason. > > Thanks a lot > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- # Jesse Molina # Mail = [EMAIL PROTECTED] # Page = [EMAIL PROTECTED] # Cell = 1.602.323.7608 # Web = http://www.opendreams.net/jesse/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote Office, Centrally Shared Voicemail
Why not simply store voicemail local so there are no issues if the VPN
goes down. Then set up your dial plan at each site to allow the PSTN
access to your remote (other site) extensions. You can use IAX to trunk
a "PSTN" call just like you can a local caller, just give them access to
the context.
Example:
Local Server
My Extension is 2995
Voicemail everything is local
Remote Server
PSTN dial in 731-555-2995
In extension.conf
[call_2000]
; This context establishes 4 digit dialing to the 2000 block
exten => _2XXX,1,Dial(IAX2/local server/${EXTEN},30,r)
exten => _2XXX,2,Hangup()
exten => _2XXX,102,Congestion()
When someone calls in and enters the 2995 extension this will route the
call through the IAX trunk.
Hope this helps.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Yingling
Sent: Friday, November 30, 2007 1:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Remote Office, Centrally Shared Voicemail
Hi,
I'm trying to set up a remote office with its own Asterisk Server
they'll have a dedicated land line, but we'll still want them connected
to the main office via VOIP (IAX2 via VPN). I've tested using IAX2 to
bridge between the offices based on extensions, since the extensions we
want to share are in isolated blocks of numbers. I'm not sure how to
handle voicemail though.
I'd like to link the voicemail so that local calls to either office will
call extensions and leave voicemail with the appropriate parties. I'd
like to avoid "Please call a new number" messages. I have some ideas:
1. Use central network storage for both offices - if the remote VPN
goes down, the remote office can't connect to the voicemail storage, so
they can't see old voicemail, and may lose new voicemail.
2. Use local storage for all voicemail. Only the local office can see
or receive voicemail. This would require a "Please call a new number"
message, I think.
3. Implement some sort of backup script - use local storage for each
office, then periodically sync voicemail folders over the VPN.
Can anyone suggest an approach to this problem?
Thanks!
Matthew Yingling
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[asterisk-users] Remote Office, Centrally Shared Voicemail
Hi, I'm trying to set up a remote office with its own Asterisk Server they'll have a dedicated land line, but we'll still want them connected to the main office via VOIP (IAX2 via VPN). I've tested using IAX2 to bridge between the offices based on extensions, since the extensions we want to share are in isolated blocks of numbers. I'm not sure how to handle voicemail though. I'd like to link the voicemail so that local calls to either office will call extensions and leave voicemail with the appropriate parties. I'd like to avoid "Please call a new number" messages. I have some ideas: 1. Use central network storage for both offices - if the remote VPN goes down, the remote office can't connect to the voicemail storage, so they can't see old voicemail, and may lose new voicemail. 2. Use local storage for all voicemail. Only the local office can see or receive voicemail. This would require a "Please call a new number" message, I think. 3. Implement some sort of backup script - use local storage for each office, then periodically sync voicemail folders over the VPN. Can anyone suggest an approach to this problem? Thanks! Matthew Yingling ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with Asterisk 1.4.14 and Queue app
It´s very strange, when Asterisk 1.4.15 crash don´t make a core file...
I´m sure it´s running with -g option!!
On Nov 30, 2007 11:02 AM, equis software <[EMAIL PROTECTED]> wrote:
> Hi, Jared. I'm going to test in 1.4.15 and then I'll tell you what
> happend.
>
> Thanks
>
>
> On Nov 29, 2007 3:35 PM, equis software <[EMAIL PROTECTED] > wrote:
>
> > You are right!
> > Here there is the backtrace
> >
> > (gdb) bt
> > #0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
> > #1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
> > "default", interpclass=0x0) at res_musiconhold.c:646
> > #2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 > out of bounds>, interpclass=0x2e0 ) at
> > channel.c:4609
> > #3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
> > app_queue.c:3600
> > #4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
> > context=0x821d040 "my-queue", exten=0x821d090 "80", priority=8, label=0x0,
> > callerid=0x819dc90 "490814", action=E_MATCHMORE) at pbx.c:532
> > #5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
> > #6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
> > #7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
> > #8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
> > #9 0xb7e3e1ba in clone () from /lib/libc.so.6
> >
> >
> > (gdb) bt full
> > #0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
> > No symbol table info available.
> > #1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
> > "default", interpclass=0x0) at res_musiconhold.c:646
> > mohclass = (struct mohclass *) 0x2e0
> > #2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 > out of bounds>, interpclass=0x2e0 ) at
> > channel.c:4609
> > No locals.
> > #3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
> > app_queue.c:3600
> > makeannouncement = 0
> > res = 136429000
> > ringing = 0
> > lu = (struct ast_module_user *) 0x8250230
> > user_priority = 0x821bdc8 "1196248345.116"
> > max_penalty_str = 0x821bdc8 "1196248345.116"
> > prio = 0
> > max_penalty = 0
> > reason = QUEUE_UNKNOWN
> > tries = 0
> > noption = 0
> > args = {argc = 5, argv = 0xb71a3908, queuename = 0xb71a3710
> > "my-queue", options = 0xb71a371b "t", url = 0xb71a371d "",
> > announceoverride = 0xb71a371e "", queuetimeoutstr = 0xb71a371f "300",
> > agi = 0x0}
> > qe = {parent = 0x8227198, moh = "default", '\0' > times>, announce = '\0' , context = '\0' > times>,
> > digits = '\0' , valid_digits = 0, pos = 1, prio = 0,
> > last_pos_said = 0, last_periodic_announce_time = 1196248351,
> > last_periodic_announce_sound = 0, last_pos = 0, opos = 1, handled = 0,
> > max_penalty = 0, start = 1196248351, expire = 1196248651, chan = 0x821cec0,
> > next = 0x0}
> > #4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
> > context=0x821d040 "my-queue", exten=0x821d090 "80", priority=8, label=0x0,
> > callerid=0x819dc90 "490814", action=E_MATCHMORE) at pbx.c:532
> > e = (struct ast_exten *) 0x8255fc0
> > res = 0
> > q = {incstack = {0x0 }, stacklen = 0, status
> > = 5, swo = 0x0, data = 0x0, foundcontext = 0x821d040 "my-queue"}
> > passdata = "my-queue|t|||300", '\0'
> > matching_action = 136488696
> > #5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
> > dst_exten = '\0' ,
> > "1¶Þ·\000\000\000\000\000\000\000\000\216\227ñ·", '\0' , "
> > Êé· Êé·D\236\032·ôßñ·
> > Êé·\f\000\000\000D\236\032·t\222ñ·0Êé·ô¯é·|\236\032·¥²Þ·
> > Êé·ôßñ·\200\026%\b¨x\024\b|\236\032·\215añ·\020\000\000\000\f\000\000\000pÍ%\b°Ò!\bÐL\"\b\000\000\000\000!\224ñ·\201
> > \006\b"
> >
> > pos = 0
> > digit = 0
> > found = 1
> > res = 0
> > error = 0
> > #6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
> > No locals.
> > #7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
> > _buffer = {__routine = 0x8067ef0 , __arg
> > = 0x1b8019, __canceltype = -1222992148, __prev = 0x0}
> > ret = (void *) 0x8224cd0
> > ---Type to continue, or q to quit---
> > a = {start_routine = 0x80c78e0 , data = 0x821cec0,
> > name = 0x8224cd0 "pbx_thread", ' ' , "started at [
> > 2632] pbx.c ast_pbx_start()"}
> > #8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
> > No symbol table info available.
> > #9 0xb7e3e1ba in clone () from /lib/libc.so.6
> > No symbol table info available.
> >
> >
> > Thanks
> >
> >
> >
> >
> >
> >
> > On Nov 29, 2007 3:04 PM, Jared Smith <[EMAIL PROTECTED]> wrote:
> >
> > > On Thu, 2007-11-29 at 14:28 -0300, equis software wrote:
> > > > I have problems with 1.4.14, it crash every few minutes.
> > > > The same configuration and machine in Asterisk 1.4.6 it doesn´t
> > > > happend.
> > >
> > > Are you able to get a good backtrace from the core file generated by
> > > the
> > > crash? Without
[asterisk-users] How to setup redundant SIP peers
Hello list,
I try to setup an asterisk-server with different SIP-Peers to PSTN.
The Peer are working and configured in sip.conf:
[peer1]
type=peer
host=10.10.10.1
[peer2]
type=peer
host=10.10.10.2
Now dialout is no problem. Extensions.conf says:
exten => _0Z.,1,Dial(SIP/49${EXTEN:[EMAIL PROTECTED],30)
But how can I setup a failure-route if the SIP-Proxy "peer1" ist not
answering (in 3sec) or send "50x" error?
Next idea is to use both peers in round-robin, if they are working.
Could someone help?
Regards
Thomas
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Re: [asterisk-users] IAX complaints? What are they?
Daryl G. Jurbala wrote: > How recent? I tried switching from 1.2 to 1.4 about 4 months ago, and > asterisk would stop accepting IAX connections in less than a day and > would need to be restarted. It has been a continuously worked on task (ever since a few months ago). Russell Bryant and others have been working on it and has improved its reliability to the point of fixing most if not all of the previously outstanding issues. I recommend trying it again. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suppressing certain queue announcement voice prompts
> > > > Short of replacing a sound file with a sound file containing only > > > > a short period of silence, is there any way to suppress certain > > > > sounds from playing during queue processing by configuring for > > > > example queues.conf or other similar files? > > > > > > Which announcements are you trying to not play? > > > > queue-thankyou for instance, to name one. Or any other of the queue-* > > files in general. From time to time it can be convenient to change > the > > exact prompts played (order and contents) due to language differences > > and personal preference of the end-users. > > The question is more like what exactly do you mean with "from time to > time"? > > Anyway, your best option is probably to create one or more prompt > languages by copying the English prompts to a new directory like "en2", > "en3" and then use Set(LANGUAGE=en3) in the dialplan when you think > this is appropriate. For each of these artificial languages you can now > decide how to modify the sound files. > > Cheers, Philipp Again, very good advice thank you Philipp. And probably a very reasonable way to do this if dynamic behaviour is needed. But in my case time-to-time was meant as "every once in a while there is a particullar installation that requires this". So statically doing this is ok in my case. I'll continue with my replace-with-silence-file method for now. Thanks for the input. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Asterisk to Asterisk SIP Call Setup?
looks like something wrong with the dial plan in the extensions.conf.. i
would recommend start debug on and see the content of "full" log may be
that give some clue.
Thanks,
Vivek
On 11/30/07, Russell Brown <[EMAIL PROTECTED]> wrote:
>
>
> I have two Asterisk systems that can route to each other via a VPN with
> firewalls disabled for testing purposes.
>
> Each Server can see (tested via nmap) UDP port 5060 on the other.
>
> So... I thought that I could simply use a Dial command in Server A's
> config to place a SIP call to Server B... but it doesn't seem to work.
>
> Server A (192.168.1.33) has:
>
>exten => *136,1,Dial(SIP/[EMAIL PROTECTED],30)
>
> but whenever a user on Server A dials '*136' the call doesn't complete
> and the CLI shows:
>
>Executing [EMAIL PROTECTED]:1] Dial("SIP/112-0071f650", "
> SIP/[EMAIL PROTECTED]|30") in new stack
>-- Called [EMAIL PROTECTED]
>-- SIP/10.10.111.13-00793520 is circuit-busy
>== Everyone is busy/congested at this time (1:0/1/0)
>
> I can't see anything in Server B's logs from 192.168.1.33
>
> What am I missing?
>
> Any pointers to help me get this working?
>
> --
> Regards,
> Russell
>
> | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
> | Lady Lodge Systems | WWW Work: http://www.lls.com |
> | Peterborough, England | WWW Play: http://www.ruffle.me.uk |
>
>
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Suppressing certain queue announcement voice prompts
Hi! > > > Short of replacing a sound file with a sound file containing only a > > > short period of silence, is there any way to suppress certain sounds > > > from playing during queue processing by configuring for example > > > queues.conf or other similar files? > > > > Which announcements are you trying to not play? > > queue-thankyou for instance, to name one. Or any other of the queue-* files > in general. From time to time it can be convenient to change the exact > prompts played (order and contents) due to language differences and personal > preference of the end-users. The question is more like what exactly do you mean with "from time to time"? Anyway, your best option is probably to create one or more prompt languages by copying the English prompts to a new directory like "en2", "en3" and then use Set(LANGUAGE=en3) in the dialplan when you think this is appropriate. For each of these artificial languages you can now decide how to modify the sound files. Cheers, Philipp ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simple Asterisk to Asterisk SIP Call Setup?
I have two Asterisk systems that can route to each other via a VPN with
firewalls disabled for testing purposes.
Each Server can see (tested via nmap) UDP port 5060 on the other.
So... I thought that I could simply use a Dial command in Server A's
config to place a SIP call to Server B... but it doesn't seem to work.
Server A (192.168.1.33) has:
exten => *136,1,Dial(SIP/[EMAIL PROTECTED],30)
but whenever a user on Server A dials '*136' the call doesn't complete
and the CLI shows:
Executing [EMAIL PROTECTED]:1] Dial("SIP/112-0071f650", "SIP/[EMAIL
PROTECTED]|30") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/10.10.111.13-00793520 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
I can't see anything in Server B's logs from 192.168.1.33
What am I missing?
Any pointers to help me get this working?
--
Regards,
Russell
| Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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Re: [asterisk-users] Fw: Remove a TDM Card
Sasa wrote: > "Tzafrir Cohen" wrote: > >> New: >> loadzone=it >> defaultzone=it >> span=1,1,3,ccs,ami >> bchan=1,2 >> dchan=3 >> span=2,1,3,ccs,ami >> bchan=4-6 >> dchan=6 >> >>> ..in zapata.conf I have: >> ; new part: >> switchtype=euroisdn >> signalling = bri_net >> priindication=outofband >> group = 1 >> channel => 1-2 >> group = 2 >> channel => 4-5 > > ..therefore I must only modify zaptel.conf and zapata.conf ?..and I don't > must unload modules ? > But when PC started without TDM card isn't a problem that is loaded > wctdm24xxp module (that is present in rc.modules and rc.modules-2.4.33.3) on > boot ? > Thanks. > I think there's been a breakdown in terminology. You do not need to unload the modules (rmmod wctdm24xxp). However, it sounds like you are using Slackware, you should (but it won't hurt anything if you don't) remove the modprobe wctdm24xxp line from your rc.modules file. If you do not remove it the modprobe will fail because the card cannot be found but the only result is maybe an error message on boot up. -Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Do While loop
You can try something like this:
exten => _X.,1,SET(condition=${RAND(1,2)})
exten => _X.,2,GotoIf($[${condition} = '1']?1:3)
exten => _X.,3,SET(Result is 2)
Regards,
Ricardo Carvalho.
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Re: [asterisk-users] Suppressing certain queue announcement voice prompts
> [EMAIL PROTECTED] wrote: > >> [EMAIL PROTECTED] wrote: > >>> Short of replacing a sound file with a sound file containing only a > >>> short period of silence, is there any way to suppress certain > sounds > >>> from playing during queue processing by configuring for example > >>> queues.conf or other similar files? > >> Which announcements are you trying to not play? > > > > queue-thankyou for instance, to name one. Or any other of the queue-* > > files in general. From time to time it can be convenient to change > the > > exact prompts played (order and contents) due to language differences > > and personal preference of the end-users. > > > > We're doing this now by replacing them with silence but I'm just > > thinking that it would be more elegant to have Asterisk not attempt > to > > play them in the first place. We've also removed the files in some > > instances but that's even worse from my point of view because then we > > get file-not-present warnings. > > The sounds used are configurable in queues.conf. For instance, if you > wanted to change queue-thankyou to play something else, you could add > the line > > queue-thankyou = mythankyoufile > > inside a queue context. Unfortunately, the order the files are played > in is not configurable. If you don't want sounds played at all, then > there are certain options which you can simply not set inside a queue > in order to not have the sounds play. If you don't set a periodic- > announce-frequency, then periodic announcements will not play. > Similarly, if you do not set an announce-frequency, then > position/holdtime announcements will not be played. Well described and I understand that perfectly. The orignal point however was if it is possible to tell the queue application to not bother with certain announcements. I was hunting for some configuration options that are either not present in the queues.conf sample file or perhaps that I could find this in some totally different file that I may not have thought of already. Not because it's unclear how to replace them (as you described very well) with for instance a file containing very short silence or configure the queue so that they are not applicable (like the periodic announcement), but just to not spend time and resources on playing a file that we would rather not hear. Thank you for your clear reply though, you make an excellent point regarding the existing configuration options. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: Remove a TDM Card
"Tzafrir Cohen" wrote: > New: > loadzone=it > defaultzone=it > span=1,1,3,ccs,ami > bchan=1,2 > dchan=3 > span=2,1,3,ccs,ami > bchan=4-6 > dchan=6 > >> >> ..in zapata.conf I have: > ; new part: > switchtype=euroisdn > signalling = bri_net > priindication=outofband > group = 1 > channel => 1-2 > group = 2 > channel => 4-5 ..therefore I must only modify zaptel.conf and zapata.conf ?..and I don't must unload modules ? But when PC started without TDM card isn't a problem that is loaded wctdm24xxp module (that is present in rc.modules and rc.modules-2.4.33.3) on boot ? Thanks. -- Salvatore. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suppressing certain queue announcement voice prompts
[EMAIL PROTECTED] wrote: >> [EMAIL PROTECTED] wrote: >>> Short of replacing a sound file with a sound file containing only a >>> short period of silence, is there any way to suppress certain sounds >>> from playing during queue processing by configuring for example >>> queues.conf or other similar files? >> Which announcements are you trying to not play? > > queue-thankyou for instance, to name one. Or any other of the queue-* files > in general. From time to time it can be convenient to change the exact > prompts played (order and contents) due to language differences and personal > preference of the end-users. > > We're doing this now by replacing them with silence but I'm just thinking > that it would be more elegant to have Asterisk not attempt to play them in > the first place. We've also removed the files in some instances but that's > even worse from my point of view because then we get file-not-present > warnings. The sounds used are configurable in queues.conf. For instance, if you wanted to change queue-thankyou to play something else, you could add the line queue-thankyou = mythankyoufile inside a queue context. Unfortunately, the order the files are played in is not configurable. If you don't want sounds played at all, then there are certain options which you can simply not set inside a queue in order to not have the sounds play. If you don't set a periodic-announce-frequency, then periodic announcements will not play. Similarly, if you do not set an announce-frequency, then position/holdtime announcements will not be played. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
How recent? I tried switching from 1.2 to 1.4 about 4 months ago, and asterisk would stop accepting IAX connections in less than a day and would need to be restarted. This is with about 50 to 100 calls at a time on each box for about 10 or 12 hours a day. Less for the other half. And all IAX calls are being passed on to a far end terminator via SIP. I was going to scrap IAX entirely because it didn't seem to scale well (for non-trunking apps, at least), but many customers need it for various reasons. Daryl On Nov 30, 2007, at 8:52 AM, zoa wrote: > IAX had some stability issues in the past, the recent releases have a > lot of iax2 fixes and should no longer have those issues. > > Zoa ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Do While loop
Hi, Is there a way to have a Do-While sort of loop, as opposed to a simple While? I have a condition that the loop depends on even for the first iteration, as it often happens in life. Regards, Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suppressing certain queue announcement voice prompts
> [EMAIL PROTECTED] wrote: > > Short of replacing a sound file with a sound file containing only a > > short period of silence, is there any way to suppress certain sounds > > from playing during queue processing by configuring for example > > queues.conf or other similar files? > > Which announcements are you trying to not play? queue-thankyou for instance, to name one. Or any other of the queue-* files in general. From time to time it can be convenient to change the exact prompts played (order and contents) due to language differences and personal preference of the end-users. We're doing this now by replacing them with silence but I'm just thinking that it would be more elegant to have Asterisk not attempt to play them in the first place. We've also removed the files in some instances but that's even worse from my point of view because then we get file-not-present warnings. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.15 crash without generating core file
Hi, I'm testing Asterisk 1.4.15 with the -g option. When it crash didn´t generate core file in the /tmp folder. What is happening?? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Off-Topic: Avaya
This is both a hardware and software licensing issue. Avaya offers a SIP server separate from their main VoIP gateway. The core platform uses H.323. Either SIP or H.323 has a license cost per registered device. We have an Avaya S8300 Communications Manager providing H.323 and have this tied to an Asterisk deployment on a Sun Microsystems server. The connection between the two systems are handled by both T1, (PRI using Qsig), and H.323. The BIG issue we have is we cannot light the message waiting light on the Avaya 46XX phones registered to the Avaya server but using Asterisk voice mail. If anyone can help we would pay to solve this. Our Asterisk is 1.2.xx. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Cabrera Obed Sent: Friday, November 30, 2007 7:30 AM To: Asterisk Users Mailing List Subject: [asterisk-users] Off-Topic: Avaya Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Suppressing certain queue announcement voice prompts
[EMAIL PROTECTED] wrote: > Short of replacing a sound file with a sound file containing only a > short period of silence, is there any way to suppress certain sounds > from playing during queue processing by configuring for example > queues.conf or other similar files? Which announcements are you trying to not play? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP Trunk and increasing volume level on diguim card
In zapata.conf you can add rxgain and txgain settings and use ztmonitor to get it set. There are some more details on this on voip-info.org. On Nov 29, 2007 1:49 AM, bilal ghayyad <[EMAIL PROTECTED]> wrote: > Hi All; > > I have an IP Trunk established between Asterisk and > the VoIP service provider, when call from my mobile to > the PBX and then enter the destination number to call > via the VoIP, I got a connection but the voice level > volume need to be increased, I am trying to find if > zaptel (diguim card) can increase the volume (if there > is any command can do that)? And if that volume level > is possible to be applied only for that IP Trunk and > not for others. > > Any Help? > Regards > Bilal > > > > > Get easy, one-click access to your favorites. > Make Yahoo! your homepage. > http://www.yahoo.com/r/hs > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
The jitter buffer is actually the same. Zoa Dr. Michael J. Chudobiak wrote: > randulo wrote: > >> On Nov 30, 2007 1:40 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: >> >>> solved these issues. I think trunking (one of the main selling points >>> of IAX due to less overhead) may be a common denominator. >>> >> That does tend to explain why I've never experienced (or at least >> noticed) problems. I never trunk which is, as you state, another >> important advantage of IAX. >> > > I find the audio quality to be better on IAX - better jitter buffer! > > I don't trunk. > > - Mike > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nov 28, 2007 Asterisk Poll Results
The poll is still open here: http://food4wine.ning.com/poll Here is a CSV file of the 99 answers. http://voipusersconference.org/poll/ There is also an XML version, but it was created by Excel so I don't know if it's worth dealing with: http://voipusersconference.org/poll/results.xml Because I screwed up (mea culpa, we're all human, or almost) there were less answers now than before, but there is more info. I'll talk about the results on the conference later today. Conference today at Noon EST: http://voipusersconference.org I'd like to try more of these short polls on focused topics like the IAX question I asked. Your suggestions are welcome. /r ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with Asterisk 1.4.14 and Queue app
Hi, Jared. I'm going to test in 1.4.15 and then I'll tell you what happend.
Thanks
On Nov 29, 2007 3:35 PM, equis software <[EMAIL PROTECTED]> wrote:
> You are right!
> Here there is the backtrace
>
> (gdb) bt
> #0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
> #1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
> "default", interpclass=0x0) at res_musiconhold.c:646
> #2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 of bounds>, interpclass=0x2e0 ) at channel.c
> :4609
> #3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
> app_queue.c:3600
> #4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
> context=0x821d040 "my-queue", exten=0x821d090 "80", priority=8, label=0x0,
> callerid=0x819dc90 "490814", action=E_MATCHMORE) at pbx.c:532
> #5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
> #6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
> #7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
> #8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
> #9 0xb7e3e1ba in clone () from /lib/libc.so.6
>
>
> (gdb) bt full
> #0 0xb7df0231 in strcasecmp () from /lib/libc.so.6
> No symbol table info available.
> #1 0xb7cc4a3f in local_ast_moh_start (chan=0x821cec0, mclass=0xb71a3788
> "default", interpclass=0x0) at res_musiconhold.c:646
> mohclass = (struct mohclass *) 0x2e0
> #2 0x080834e5 in ast_moh_start (chan=0x64, mclass=0x64 of bounds>, interpclass=0x2e0 ) at channel.c
> :4609
> No locals.
> #3 0xb73eb17b in queue_exec (chan=0x821cec0, data=0xb71a3788) at
> app_queue.c:3600
> makeannouncement = 0
> res = 136429000
> ringing = 0
> lu = (struct ast_module_user *) 0x8250230
> user_priority = 0x821bdc8 "1196248345.116"
> max_penalty_str = 0x821bdc8 "1196248345.116"
> prio = 0
> max_penalty = 0
> reason = QUEUE_UNKNOWN
> tries = 0
> noption = 0
> args = {argc = 5, argv = 0xb71a3908, queuename = 0xb71a3710
> "my-queue", options = 0xb71a371b "t", url = 0xb71a371d "",
> announceoverride = 0xb71a371e "", queuetimeoutstr = 0xb71a371f "300",
> agi = 0x0}
> qe = {parent = 0x8227198, moh = "default", '\0' times>, announce = '\0' , context = '\0' times>,
> digits = '\0' , valid_digits = 0, pos = 1, prio = 0,
> last_pos_said = 0, last_periodic_announce_time = 1196248351,
> last_periodic_announce_sound = 0, last_pos = 0, opos = 1, handled = 0,
> max_penalty = 0, start = 1196248351, expire = 1196248651, chan = 0x821cec0,
> next = 0x0}
> #4 0x080c5d6b in pbx_extension_helper (c=0x821cec0, con=0x64,
> context=0x821d040 "my-queue", exten=0x821d090 "80", priority=8, label=0x0,
> callerid=0x819dc90 "490814", action=E_MATCHMORE) at pbx.c:532
> e = (struct ast_exten *) 0x8255fc0
> res = 0
> q = {incstack = {0x0 }, stacklen = 0, status =
> 5, swo = 0x0, data = 0x0, foundcontext = 0x821d040 "my-queue"}
> passdata = "my-queue|t|||300", '\0'
> matching_action = 136488696
> #5 0x080c6a31 in __ast_pbx_run (c=0x821cec0) at pbx.c:2293
> dst_exten = '\0' ,
> "1¶Þ·\000\000\000\000\000\000\000\000\216\227ñ·", '\0' , "
> Êé· Êé·D\236\032·ôßñ·
> Êé·\f\000\000\000D\236\032·t\222ñ·0Êé·ô¯é·|\236\032·¥²Þ·
> Êé·ôßñ·\200\026%\b¨x\024\b|\236\032·\215añ·\020\000\000\000\f\000\000\000pÍ%\b°Ò!\bÐL\"\b\000\000\000\000!\224ñ·\201
> \006\b"
>
> pos = 0
> digit = 0
> found = 1
> res = 0
> error = 0
> #6 0x080c78f1 in pbx_thread (data=0x64) at pbx.c:2608
> No locals.
> #7 0x080f7759 in dummy_start (data=0x64) at utils.c:843
> _buffer = {__routine = 0x8067ef0 , __arg =
> 0x1b8019, __canceltype = -1222992148, __prev = 0x0}
> ret = (void *) 0x8224cd0
> ---Type to continue, or q to quit---
> a = {start_routine = 0x80c78e0 , data = 0x821cec0,
> name = 0x8224cd0 "pbx_thread", ' ' , "started at [
> 2632] pbx.c ast_pbx_start()"}
> #8 0xb7f1513d in pthread_start_thread () from /lib/libpthread.so.0
> No symbol table info available.
> #9 0xb7e3e1ba in clone () from /lib/libc.so.6
> No symbol table info available.
>
>
> Thanks
>
>
>
>
>
>
> On Nov 29, 2007 3:04 PM, Jared Smith <[EMAIL PROTECTED]> wrote:
>
> > On Thu, 2007-11-29 at 14:28 -0300, equis software wrote:
> > > I have problems with 1.4.14, it crash every few minutes.
> > > The same configuration and machine in Asterisk 1.4.6 it doesn´t
> > > happend.
> >
> > Are you able to get a good backtrace from the core file generated by the
> > crash? Without more details, it's going to be close to impossible for
> > the Asterisk developers to guess at why it's crashing for you.
> >
> > There's some good information at
> > http://www.asterisk.org/doxygen/1.4/AstDebug.html on how to generate the
> > backtrace and attach it to a bug in the bug tracker.
> >
> >
> > --
> > Jared Smith
> > Community Relations Manager
> > Digium, Inc.
> >
> >
> > __
Re: [asterisk-users] IAX complaints? What are they?
IAX had some stability issues in the past, the recent releases have a lot of iax2 fixes and should no longer have those issues. Zoa Steve Totaro wrote: > randulo wrote: > >> Hi, >> >> We all know what the principal advantage of IAX is, doing it all on a >> single port, right? But now and again I hear complaints about it. What >> specific griefs have you had with IAX and has it stopped you from >> using it entirely? Under what conditions have you had problems? >> >> I have used SIP and IAX for about three years now. We don't do a lot >> of traffic, but I haven't really seen a difference in quality or >> dropped calls. >> >> What have others on the list experienced? >> >> tia >> >> randy >> >> > > I am not sure why, what versions, under what conditions, but audio > cutting out has been seen many times. Simply switching to SIP has > solved these issues. I think trunking (one of the main selling points > of IAX due to less overhead) may be a common denominator. > > Thanks, > Steve > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
Same with me IAX trunking worked great up until about 10 calls. Then it went down hill. This was back on 1.2, I haven't tried it since. So maybe it has been fixed? Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of randulo Sent: Friday, November 30, 2007 8:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX complaints? What are they? On Nov 30, 2007 1:40 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > solved these issues. I think trunking (one of the main selling points > of IAX due to less overhead) may be a common denominator. That does tend to explain why I've never experienced (or at least noticed) problems. I never trunk which is, as you state, another important advantage of IAX. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Off-Topic: Avaya
Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
randulo wrote: > On Nov 30, 2007 1:40 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: >> solved these issues. I think trunking (one of the main selling points >> of IAX due to less overhead) may be a common denominator. > > That does tend to explain why I've never experienced (or at least > noticed) problems. I never trunk which is, as you state, another > important advantage of IAX. I find the audio quality to be better on IAX - better jitter buffer! I don't trunk. - Mike ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Suppressing certain queue announcement voice prompts
Short of replacing a sound file with a sound file containing only a short period of silence, is there any way to suppress certain sounds from playing during queue processing by configuring for example queues.conf or other similar files? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
On Nov 30, 2007 1:40 PM, Steve Totaro <[EMAIL PROTECTED]> wrote: > solved these issues. I think trunking (one of the main selling points > of IAX due to less overhead) may be a common denominator. That does tend to explain why I've never experienced (or at least noticed) problems. I never trunk which is, as you state, another important advantage of IAX. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
randulo wrote: > Hi, > > We all know what the principal advantage of IAX is, doing it all on a > single port, right? But now and again I hear complaints about it. What > specific griefs have you had with IAX and has it stopped you from > using it entirely? Under what conditions have you had problems? > > I have used SIP and IAX for about three years now. We don't do a lot > of traffic, but I haven't really seen a difference in quality or > dropped calls. > > What have others on the list experienced? > > tia > > randy > I am not sure why, what versions, under what conditions, but audio cutting out has been seen many times. Simply switching to SIP has solved these issues. I think trunking (one of the main selling points of IAX due to less overhead) may be a common denominator. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newb Question
On 11/30/07, Vivek Shrivastava <[EMAIL PROTECTED]> wrote: > > you can try Cain & Abel ( to route calls) and Wireshark to record all the > calls. > > On 11/29/07, Adam Moffett <[EMAIL PROTECTED]> wrote: > > > > I'm pretty sure asterisk won't do that without modification. You'll > > need to do packet sniffing and decode the datathere may be products > > that do this, but asterisk is not it. > > > > And we're assuming the calls are unencrypted? > > > I inherited an office with phones that are hosted off-site. Everything > > > is skinny and G729. I see that the FreeBSD asterisk port comes with a > > > G729 codec. > > > > > > I want to record everything. If I use port mirroring on my switch, is > > > it possible to configure asterisk to record and assemble packets that > > > it doesn't otherwise route? Is it insane to user asterisk for this > > > purpose? Advice or a link to a howto would be greatly appreciated. > > > > > > > > > > > > > Thanks everyone. You can indeed use cain and abel to convert g729 to .wav (wireshark doesn't have that codec just yet), and it's easy enough to capture packets with tcpdump or wireshark. I've done this a few times as an experiment. >I suspect either you want to insert an Asterisk system in-between as a >"tap" (requiring re-configuring your phones and your outside provider) or >using a "voip sniffer" plugged into the management port of your Ethernet >switch. That's more or less it. I know how to duplicate the packets. What I want now is to automatically reassemble, decode and archive each rtp stream (every call) on this network of 20 users or so. There are open source applications for this but they don't have G729 support. Asterisk has G729 support and it can record calls (at least from the command line it appears to have that feature). All the pieces are there but I'm not even sure where to begin configuring it to do only that. Surely some adventurous soul has done this already. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: Remove a TDM Card
On Wed, Nov 28, 2007 at 04:59:22PM +0100, Sasa wrote: > Hi, sorry but perhaps I don't have explained clearly my problem...now I have > a box voip that must be replace with another box voip but I want to do test > before remove the old voip from production. > > The box voip (named 1) that now is in production have three card, two isdn > card and TDM2400P that I want remove for to install in the new box voip > (named 2). > > On the box voip 1 I have: > > Asterisk version 1.2.13 > The kernel version is: 2.6.19.2 > > ..but on the box voip 2 I have the new asterisk version and kernel. > > On box voip 1 I have: > > zaptel.conf: > loadzone=it > defaultzone=it > span=2,1,3,ccs,ami > bchan=25-26 > dchan=27 > span=3,1,3,ccs,ami > bchan=28-29 > dchan=30 > fxsls=1-24 New: loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1,2 dchan=3 span=2,1,3,ccs,ami bchan=4-6 dchan=6 > > ..in zapata.conf I have: > [channels] > language=it > ... > ... > ;Linee ISDN > immediate=no > switchtype=euroisdn > signalling = bri_net > priindication=outofband > group = 1 > channel => 25-26 > group = 2 > channel => 28-29 ; new part: switchtype=euroisdn signalling = bri_net priindication=outofband group = 1 channel => 1-2 group = 2 channel => 4-5 > > ;Linee tdm > immediate=yes > .. > .. > cidstart=ring > signalling=fxs_ls > group = 3 > channel => 1-5 ; And get rid of that. ; immediate=yes ??? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: Remove a TDM Card
On Thu, Nov 29, 2007 at 11:14:12AM +0100, Sasa wrote: > Hi, my problem isn't on new voip box with latest asterisk version...my > problem is on voip with Asterisk 1.2.13 where I must remove TDM Card, this > steps for remove rightly TDM Card: > > - remove line configuration about tdm card in zapata.conf and zaptel.conf > - remove in rc.modules and rc.modules-2.4.33.3 line: > /sbin/modprobe wctdm24xxp && /sbin/ztcfg -vv > - rmmod wctdm24xxp > - halt > - remove physically card tdm from pc (box voip 1) > - restart box voip 1 > > ..this procedure is ok ? > Thanks ! Totally wrong. No need to unload the modules before you shut down the system. No need to "deconfigure" them just before you shut down just because you happen to be removing them later. What you need to do: edit /etc/zaptel.conf and /etc/asterisk/zapata.conf to match your new configuration , shut down the box, take away the card and boot again. The proper configuration, in a different page. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT - Which TAPI driver to use ?
Hi, I'm trying to use latest versions of ActivaTSP and Asttapi with an Astmanproxy-enabled 1.4 Asterisk. Up to now, I can't find a way to teach Outlook 2002 how to use any of those TAPI drivers: when using "Call this contact" in Outlook Contacts pane, I can't see and select any TAPI driver. Beside that, I'm wondering how those software compare and if they are usable today. >From googling, here are my findings : - ActivaTSP can't work with Astmanproxy as Asmanproxy needs to be patched, - Asttapi wouldn't terminate a completed call. Which option would you pick ? Is there any other option (free or commercial) for Outlook click2call ? Best regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: Remove a TDM Card
"Tzafrir Cohen" wrote: > You have been quite short on details. For instance: what distribution of > Linux? What version of Zaptel? > > Do you have another Zaptel card? It seems you either have two zaphfc > cards or one dual-BRI card. If so, the procedure is slightly more > complicated, as you basically have to reconfigure the system afterwards. > > As I mentioned, genzaptelconf can be handy for that. I don't know what Linux distribution is installed but the kernel version is 2.6.19.2, the zaptel version is zaptel-1.2.12 and is present one TDM Card and two zaphfc cards..with this architecture is correct my procedure for remove TDM card ? Thanks. -- Salvatore. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX complaints? What are they?
randulo wrote: > What > specific griefs have you had with IAX and has it stopped you from > using it entirely? With SIP you can "attach" custom variables to calls (using X-... headers). IAX (Inter-Asterisk eXchange!) can't do that (yet). Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to originate a call from console CLI ?
yup with chan_oss On 11/30/07, Olivier <[EMAIL PROTECTED]> wrote: > > > 2007/11/30, Vivek Shrivastava <[EMAIL PROTECTED]>: > > > > I am not sure if this fits in your requirement but try "dial" command. > > > > Do you mean, dialing both extensions one after the other and then, bridge > them ? > Or do you mean using the asterisk Chan_OSS capabilities ? > > Cheers > > > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hylafax
email the biz list. you should get some one there. - Original Message - From: "Sahil Gupta" <[EMAIL PROTECTED]> To: Sent: Thursday, November 29, 2007 1:32 PM Subject: [asterisk-users] Hylafax > Hi, > We seem to be having some teething issues with a new Hylafax - happy to > pay > someone to complete the installation. Please contact offlist. > > Regards, > > > Sahil Gupta > Chief Executive Officer > VoiceValley Group of Companies > > Phone: +61-7-30188403 > Fax: +61-7-30188499 > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX complaints? What are they?
Hi, We all know what the principal advantage of IAX is, doing it all on a single port, right? But now and again I hear complaints about it. What specific griefs have you had with IAX and has it stopped you from using it entirely? Under what conditions have you had problems? I have used SIP and IAX for about three years now. We don't do a lot of traffic, but I haven't really seen a difference in quality or dropped calls. What have others on the list experienced? tia randy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial.
- Original Message - From: "Tilghman Lesher" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Saturday, November 24, 2007 5:33 PM Subject: Re: [asterisk-users] Asterisk+HylaFAX+SpanDSP+IAXmodem tutorial. > On Saturday 24 November 2007 00:16:11 Steve Totaro wrote: >> Alex Balashov wrote: >> > Asterisk 1.4 does have this ability natively. However, it is somewhat >> > limited in its flexibility / in terms of what I can do with it, and >> > I have gotten reports that HylaFAX works better. I haven't actually >> > done a comparison between the two. >> > >> > Being someone who hates 1.2, I was strongly tempted to go this route, >> > though. >> >> Why would anyone hate the most stable version of Asterisk? >> >> What is ABE using these days? If it is not 1.4, I wonder why? Maybe so >> all the free developers and eager and silly early adopters can iron out >> the bugs, submit patches and sign away their rights. I am sure if they >> are not using 1.4 it probably has something to do with reliability and >> the costs of supporting that release. Any other theories? > > Yeah, that version C is currently in beta and is very close to release. > ABE > has to be put through its paces before release and that takes time. I'm > sorry > if that seems like evidence that Digium isn't supporting 1.4, but it > simply > isn't true. > > -- > Tilghman > Quick question. How long has 1.4.X been out ? Sitll no ABE ? We have been told that ABE for 1.4.X will be out shortly for a lil while already ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to originate a call from console CLI ?
Brillant !! I don't know why, I wanted to substitute extension keyword with a value. Thanks for the tip. 2007/11/30, Philipp Kempgen <[EMAIL PROTECTED]>: > > Olivier wrote: > > > Usage2: originate extension [EMAIL PROTECTED] > > > I would like for example to call 0123456789 number from SIP/7530 > extension. > > My asterisk server is set to use "local" context for outgoing calls. > > My first idea was to type this : > > originate SIP 7530 [EMAIL PROTECTED] > > How about > originate SIP/7530 extension [EMAIL PROTECTED] > ? > > Regards, > Philipp Kempgen > > -- > amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de > Let's use IT to solve problems and not to create new ones. > Asterisk? -> http://www.das-asterisk-buch.de > > Geschäftsführer: Stefan Wintermeyer > Handelsregister: Neuwied B 14998 > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to originate a call from console CLI ?
2007/11/30, Vivek Shrivastava <[EMAIL PROTECTED]>: > > I am not sure if this fits in your requirement but try "dial" command. > Do you mean, dialing both extensions one after the other and then, bridge them ? Or do you mean using the asterisk Chan_OSS capabilities ? Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to originate a call from console CLI ?
Olivier wrote: > Usage2: originate extension [EMAIL PROTECTED] > I would like for example to call 0123456789 number from SIP/7530 extension. > My asterisk server is set to use "local" context for outgoing calls. > My first idea was to type this : > originate SIP 7530 [EMAIL PROTECTED] How about originate SIP/7530 extension [EMAIL PROTECTED] ? Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Pcengines Alix board
Hi, 2007/11/30, John Faubion <[EMAIL PROTECTED]>: > > > Thanks for the tip. It seems like they no longer manufacture them: > > > > http://www.neoware.com/products/hardware/ > > No, but the Neoware e140 has a PCI expansion slot, is expandable to 1GB > RAM, > and still has room inside the case for a hard drive. It is available > without > Win XPe starting at $339 new. The prices on these are coming down. Is the PCI slot large enough for full height, half length PCI boards ? Has you heard of a PCI Express version ? Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to originate a call from console CLI ?
I am not sure if this fits in your requirement but try "dial" command. --Vivek On 11/29/07, Olivier <[EMAIL PROTECTED]> wrote: > > Hi, > > I would like to originate my first call from CLI. > As I'm new to this, I'm wondering if it's possible. > When I type "originate" from CLI, I've got this : > > " There are two ways to use this command. A call can be originated > between a > channel and a specific application, or between a channel and an extension > in > the dialplan. This is similar to call files or the manager originate > action. > Calls originated with this command are given a timeout of 30 seconds. > > Usage1: originate application [appdata] > This will originate a call between the specified channel tech/data and > the > given application. Arguments to the application are optional. If the given > > arguments to the application include spaces, all of the arguments to the > application need to be placed in quotation marks. > > Usage2: originate extension [EMAIL PROTECTED] > This will originate a call between the specified channel tech/data and > the > given extension. If no context is specified, the 'default' context will be > used. If no extension is given, the 's' extension will be used." > > > I would like for example to call 0123456789 number from SIP/7530 > extension. > My asterisk server is set to use "local" context for outgoing calls. > My first idea was to type this : > originate SIP 7530 [EMAIL PROTECTED] > > But it fails : it keeps displaying " There are two ways ..." and nothing > else seem to occur. > > Can anyone help ? > Cheers > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
