[asterisk-users] Asterisk and spandsp 0.3
Hi As there been any progress regarding the use of spandsp 0.3 with Asterisk 1.2.13? Last month there was a thread about how spandsp 0.3 and rxfax from http://www.soft-switch.org/downloads/snapshots/spandsp made asterisk crash. Is there any resources on how to get spandsp 0.3 work with Asterisk otherwise? Thank you JY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference between skinny user and many sip user
Ok thanks, do you think that it isn't possible to do that automatically from asterisk? On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I think, that adhoc conferencing isn't possible in this way, instead you should use meetme, ie.: skinny user calls to user A and transfer his to meetme number skinny user calls to user B and transfer his to meetme number skinny user calls to meetme number all three speech in conference... nik600 wrote: Hi, can i set up my asterisk for: - receive a skinny call in a specific context (yes, i have already compiled asteirsk with h323 support) - forward the call to a sip user A - make the sip user B join the call and create a conference between skinny caller, A and B maky thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Input on Dundi
hi, all,
I have realized a dundi cluster,,,
the details, please read...
http://jefferychen1977.spaces.live.com/blog/cns!9E49EEC4251C4476!494.entry
Thanks,...
On 12/13/06, David Thomas [EMAIL PROTECTED] wrote:
On 12/12/06, JR Richardson [EMAIL PROTECTED] wrote:
1.) When a registration server fails there doesn't seem to be an easy
way to have clients automatically register to a new server. (our
clients are mostly other asterisk boxes.) To solve this we are
considering using DNS failover.
When registering with an Asterisk server to an Asterisk cluster of
servers, for the purpose of traversing a NAT or something else (to
solve a problem where direct contact cannot be performed), I would
suggest doing multiple registration to two registration servers, using
different names.
Like
registration [name1] to registration server 1
registration [name2] to registration server 2
in the outgoing dilaplan
exten = _NXXNXX,1,Dial(IAX2/server1..|j)
exten = _NXXNXX,102,Dial(IAX2/server2..
so if server one is not there the call will jump to the next server
or
exten = _NXXNXX,1,Dial(IAX2/server1IAX2/server2.
first server to answer will get the call.
you can do something similar calling from the cluster to the end
Asterisk server
dundi lookup for [name1] if not available lookup [name2]
2.) If you plan to do any direct routing using the fullcontact
address like what is shown in JR's whitepaper, you may find that
fullcontact sometimes contains private network addresses. This makes
it impossible to route inbound calls directly to the client.
I recently started pulling the ipaddress and port from the database
instead of using the fullcontact field. Aaron Daniels helped me to
get the realtime query working instead of using the mysql connect
statements.
[lookupmysql]
include = invalid
exten = _X.,1,RealTime(sippeers|name|${EXTEN}|DN_)
exten = _X.,2,GotoIf($[${DN_ipaddr} = ]?${EXTEN},105:${EXTEN},3)
exten = _X.,3,Set([EMAIL PROTECTED]:${DN_port})
exten = _X.,4,Dial(SIP/${directdial},15,rj)
exten = _X.,5,Macro(sendtovm,${EXTEN})
exten = _X.,6,Hangup
exten = _X.,105,Macro(sendtovm,${EXTEN})
exten = _X.,106,Hangup
The RealTime command pulls all the entire record from the database and
prepends all the fields with the last argument (here is have DN_) so
when the record is pulled, all the records info is available as a
variable like DN_port and DN_ipaddr.
This is a really cool command. Hope this helps.
Wow, thanks for the examples JR. This is exactly what I needed. I was
not aware of the RealTime command. That will be very useful.
David
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Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Matt Gibson wrote: Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. maybe some missing in your xml config file? here is my minimalistic .cnf.xml, that works for my 7961 device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword***/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentp.ujf.cas.cz/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeName192.168.0.100/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg711a/preferredCodec natEnabled0/natEnabled phoneLabelAsterisk/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 961/featureLabel proxy192.168.0.100/proxy name961/name displayNamePJ7961/displayName authName961/authName authPassword***/authPassword messagesNumber8299/messagesNumber /line line button=2 featureID21/featureID featureLabelEcho test/featureLabel speedDialNumber959/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePassword***/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp /device Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches
CDP has nothing to do with inline power, it is L2 proprietary protocol for negotiation of voice vlan between phone and switch, so you don't need to set what vlan number phone should use for voice and what is for connected pc data. if you disable cdp on switch, phone will still working, except you should set voice vlan manually through phone menu. PJ William McCloskey wrote: Layer 2 switches support all the basic switching functionality. QoS, SNMP, POE, VLANs, Etc... depending on the model and features. Layer 3 switches are essentialy basic routers with a switch built in. One thing about Cisco CDP and a lot of POE switches is you can get CDP support with a custom Ethernet cable, just swap pins 4-5 with 7-8 (This is how I'm running Cisco 7940G's with a Dell POE Switch). - William J McCloskey Information Technology Manager [EMAIL PROTECTED] 503-827-8141 503-228-6747 fax www.timbercon.com - -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Monday, December 11, 2006 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Re: Recommendations for QoS, PoE Switches Edgewater Networks markets a 24 port switch, with PoE (both Cisco CDP and 802.3af supported), and Layer 2/3 management features that retails for less than $1500. The model is EC-2402POE-01 Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick May Sent: Monday, December 11, 2006 10:22 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches On Mon, Dec 11, 2006 at 09:53:53AM -0500, Zeeshan Zakaria wrote: What's the price for these HP switches? And also I someone can give me a link to some document where I can read about Layer 2 and Layer 3, how they help in VoIP traffic, it'll be helpful. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CDW's retail price was about $7,000. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference between skinny user and many sip user
maybe some asterisk guru have idea for some smart script, how to do this ;-) I found some RFC for better sip conferencing, but currently probably not implemented in asterisk :'( High-Level Requirements for Tightly Coupled SIP Conferencing ftp://ftp.rfc-editor.org/in-notes/rfc4245.txt nik600 wrote: Ok thanks, do you think that it isn't possible to do that automatically from asterisk? On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I think, that adhoc conferencing isn't possible in this way, instead you should use meetme, ie.: skinny user calls to user A and transfer his to meetme number skinny user calls to user B and transfer his to meetme number skinny user calls to meetme number all three speech in conference... nik600 wrote: Hi, can i set up my asterisk for: - receive a skinny call in a specific context (yes, i have already compiled asteirsk with h323 support) - forward the call to a sip user A - make the sip user B join the call and create a conference between skinny caller, A and B maky thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA2100 sends an unexpected BYE message whentransmitting a FAX
Hi Mike, Do you have a full SIP trace? Cheers Dave Dave, here is the trace, the BYE message at the end: -- SIP read from 192.168.6.222:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.6.222:5060;branch=z9hG4bK-a993e341 From: 201 sip:[EMAIL PROTECTED];tag=c6bf15cfef0f4e11o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE Max-Forwards: 70 Contact: 201 sip:[EMAIL PROTECTED]:5060 Expires: 240 User-Agent: Linksys/SPA2100-3.3.6 Content-Length: 255 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 413461 413461 IN IP4 192.168.6.222 s=- c=IN IP4 192.168.6.222 t=0 0 m=audio 16384 RTP/AVP 8 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv chan_sip.c:3377 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.6.222:5060;branch=z9hG4bK-a993e341 (59) chan_sip.c:3377 parse_request: Header 2: From: 201 sip:[EMAIL PROTECTED];tag=c6bf15cfef0f4e11o0 (56) chan_sip.c:3377 parse_request: Header 3: To: sip:[EMAIL PROTECTED] (33) chan_sip.c:3377 parse_request: Header 4: Call-ID: [EMAIL PROTECTED] (40) chan_sip.c:3377 parse_request: Header 5: CSeq: 101 INVITE (16) chan_sip.c:3377 parse_request: Header 6: Max-Forwards: 70 (16) chan_sip.c:3377 parse_request: Header 7: Contact: 201 sip:[EMAIL PROTECTED]:5060 (41) chan_sip.c:3377 parse_request: Header 8: Expires: 240 (12) chan_sip.c:3377 parse_request: Header 9: User-Agent: Linksys/SPA2100-3.3.6 (33) chan_sip.c:3377 parse_request: Header 10: Content-Length: 255 (19) chan_sip.c:3377 parse_request: Header 11: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61) chan_sip.c:3377 parse_request: Header 12: Supported: x-sipura (19) chan_sip.c:3377 parse_request: Header 13: Content-Type: application/sdp (29) chan_sip.c:3377 parse_request: Header 14: (0) chan_sip.c:3409 parse_request: Line: v=0 (3) chan_sip.c:3409 parse_request: Line: o=- 413461 413461 IN IP4 192.168.6.222 (38) chan_sip.c:3409 parse_request: Line: s=- (3) chan_sip.c:3409 parse_request: Line: c=IN IP4 192.168.6.222 (22) chan_sip.c:3409 parse_request: Line: t=0 0 (5) chan_sip.c:3409 parse_request: Line: m=audio 16384 RTP/AVP 8 100 101 (31) chan_sip.c:3409 parse_request: Line: a=rtpmap:8 PCMA/8000 (20) chan_sip.c:3409 parse_request: Line: a=rtpmap:100 NSE/8000 (21) chan_sip.c:3409 parse_request: Line: a=fmtp:100 192-193 (18) chan_sip.c:3409 parse_request: Line: a=rtpmap:101 telephone-event/8000 (33) chan_sip.c:3409 parse_request: Line: a=fmtp:101 0-15 (15) chan_sip.c:3409 parse_request: Line: a=ptime:30 (10) chan_sip.c:3409 parse_request: Line: a=sendrecv (10) --- (14 headers 13 lines)--- chan_sip.c:3147 sip_alloc: Allocating new SIP dialog for [EMAIL PROTECTED] - INVITE (With RTP) chan_sip.c:11375 handle_request: Received INVITE (5) - Command in SIP INVITE chan_sip.c:1010 parse_sip_options: Begin: parsing SIP Supported: x-sipura chan_sip.c:1022 parse_sip_options: Found SIP option: -x-sipura- chan_sip.c:1033 parse_sip_options: Found no match for SIP option: x-sipura (Please file bug report!) chan_sip.c:1039 parse_sip_options: * SIP extension value: 0 for call [EMAIL PROTECTED] Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.6.222 : 5060 (non-NAT) Dec 12 12:47:49 DEBUG[7416]: chan_sip.c:7258 check_user_full: Setting NAT on RTP to 0 Reliably Transmitting (no NAT) to 192.168.6.222:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.6.222:5060;branch=z9hG4bK-a993e341;received=192.168.6.222 From: 201 sip:[EMAIL PROTECTED];tag=c6bf15cfef0f4e11o0 To: sip:[EMAIL PROTECTED];tag=as14e77426 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: insido Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=36741c67 Content-Length: 0 --- Dec 12 12:47:49 DEBUG[7416]: chan_sip.c:1299 __sip_reliable_xmit: *** SIP TIMER: Initalizing retransmit timer on packet: Id #256 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '201' ipbx100*CLI -- SIP read from 192.168.6.222:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.6.222:5060;branch=z9hG4bK-a993e341 From: 201 sip:[EMAIL PROTECTED];tag=c6bf15cfef0f4e11o0 To: sip:[EMAIL PROTECTED];tag=as14e77426 Call-ID: [EMAIL PROTECTED] CSeq: 101 ACK Max-Forwards: 70 Contact: 201 sip:[EMAIL PROTECTED]:5060 User-Agent: Linksys/SPA2100-3.3.6 Content-Length: 0 chan_sip.c:3377 parse_request: Header 0: ACK sip:[EMAIL PROTECTED] SIP/2.0 (39) Dchan_sip.c:3377 parse_request: Header 1: Via: SIP/2.0/UDP 192.168.6.222:5060;branch=z9hG4bK-a993e341 (59) chan_sip.c:3377 parse_request: Header 2: From: 201 sip:[EMAIL PROTECTED];tag=c6bf15cfef0f4e11o0 (56) chan_sip.c:3377 parse_request: Header 3: To: sip:[EMAIL PROTECTED];tag=as14e77426 (48) chan_sip.c:3377 parse_request: Header 4: Call-ID: [EMAIL
Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Hi Is NAT set to NO? It needs to be set to NO in 8.0.3 or it just sits there at registering as you say Thanks - Original Message - From: Pavel Jezek [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 13, 2006 9:08 AM Subject: Re: [asterisk-users] Cisco 7970 + New Firmware (8.2) Matt Gibson wrote: Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. maybe some missing in your xml config file? here is my minimalistic .cnf.xml, that works for my 7961 device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword***/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentp.ujf.cas.cz/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeName192.168.0.100/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg711a/preferredCodec natEnabled0/natEnabled phoneLabelAsterisk/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 961/featureLabel proxy192.168.0.100/proxy name961/name displayNamePJ7961/displayName authName961/authName authPassword***/authPassword messagesNumber8299/messagesNumber /line line button=2 featureID21/featureID featureLabelEcho test/featureLabel speedDialNumber959/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePassword***/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp /device Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long busy()
[Description]
Busy([timeout]): This application will indicate the busy condition to
the calling channel. If the optional timeout is specified, the calling
channel
will be hung up after the specified number of seconds. Otherwise, this
application will wait until the calling channel hangs up.
This is what I found when I typed show application busy in the CLI.
Did I interpret it wrong?
regards, Christophorus
Mailinglisten schrieb:
Christophorus Laube schrieb:
hi list,
I set up a new asterisk machine with asterisk 1.2.13 and misdn
0.3.1rc27.
I use an e1 card with sip clients. My extensions look like this:
[E1]
snip...snip
exten = 33006733,1,Set(CALLED=${EXTEN})
exten = 33006733,2,Dial(SIP/[EMAIL PROTECTED])
exten = 33006733-ANSWER,3,Answer()
[SIP]
exten = _X.,1,Noop()
exten = _X.,2,SetCallerPres(allowed_passed_screen)
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40)
exten = _X.-BUSY,4,Busy(1)
But whenever a sip client calls to an exten that is busy through e1 I
get busy tones for 10s before I get disconnected. But I want to have
it only for 1s.
Does anyone know how to fix that?
regards, Christophorus
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AFAIK the BUSY() command has nothing to do with the busy indication.
You can't pass anything to this command.
Check: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Busy
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[asterisk-users] Problem with asterisk 1.4 Installation (undefined reference to `ast_copy_string')
Hi. After successfully running ./configure I run make. When running make I get the following error.. [CC] ast_expr2f.c - ast_expr2f.o [CC] ast_expr2.c - ast_expr2.o [CC] strcompat.c - strcompat.o [LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o - aelparse aelparse.o(.text+0x3029): In function `ael_yylex': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ael.flex:417: undefined reference to `ast_copy_string' ast_expr2f.o(.text+0x1198): In function `ast_expr': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ast_expr2.fl:248: undefined reference to `ast_copy_string' collect2: ld returned 1 exit status make[1]: *** [aelparse] Error... plz could answer this issue. -nsthi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with asterisk 1.4 Installation (undefined reference to `ast_copy_string')
Hi. After successfully running ./configure I run make. When running make I get the following error.. [CC] ast_expr2f.c - ast_expr2f.o [CC] ast_expr2.c - ast_expr2.o [CC] strcompat.c - strcompat.o [LD] aelparse.o aelbison.o pbx_ael.o ael_main.o ast_expr2f.o ast_expr2.o strcompat.o - aelparse aelparse.o(.text+0x3029): In function `ael_yylex': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ael.flex:417: undefined reference to `ast_copy_string' ast_expr2f.o(.text+0x1198): In function `ast_expr': /usr/src/asterisk1/asterisk-1.4.0-beta3/utils/ast_expr2.fl:248: undefined reference to `ast_copy_string' collect2: ld returned 1 exit status make[1]: *** [aelparse] Error... plz could answer this issue. -nsthi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
there is any way to configure a 7970 without using the display, I have my LCD broken so I cannot see what I'm doing :) but the phone works fine. 2006/12/13, Paul A Brown [EMAIL PROTECTED]: Hi Is NAT set to NO? It needs to be set to NO in 8.0.3 or it just sits there at registering as you say Thanks - Original Message - From: Pavel Jezek [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 13, 2006 9:08 AM Subject: Re: [asterisk-users] Cisco 7970 + New Firmware (8.2) Matt Gibson wrote: Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. maybe some missing in your xml config file? here is my minimalistic .cnf.xml, that works for my 7961 device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword***/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentp.ujf.cas.cz/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeName192.168.0.100/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg711a/preferredCodec natEnabled0/natEnabled phoneLabelAsterisk/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 961/featureLabel proxy192.168.0.100/proxy name961/name displayNamePJ7961/displayName authName961/authName authPassword***/authPassword messagesNumber8299/messagesNumber /line line button=2 featureID21/featureID featureLabelEcho test/featureLabel speedDialNumber959/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePassword***/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp /device Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM04B and shared IRQ ..but asterisk can work..
Hello, I have installed asterisk version 1.2.12 and latest zaptel modules. but i can see some IRQ conflicts on the server. iam uisng two TDM04B cards. according to my previous knowledge on asterisk verison 1.07 asterisk has given lot of erros when starting if you have assigned the same IRQ number to any other device. My question is new releae version 1.2.12 has resolved the IRQ issue ?? TDM04B cards can share IRQ with another device. or should it have a unique IRQ for porper performance. any drawbacks with using the shared IRQ ? i mean random call hangups../..echo..etc ( i can start my asterisk verison even with IRQ sharing..but earlier version didnt support it..) hope new verison has fixed that issue. please let me know.. my cat /proc/inturrupts with in 10 sec gives 18 interrupts. and zttest is 99.87% problem is i cant change my PCI slots..i have only 4 slots..2 for NIC's and 2 for TDM04B's and also kernel 2.6 is preemptive... what do u all recommend ? is it always need a unique IRQ for TDM04B cards.. many thanks, Tharanga ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long busy()
Christophorus Laube schrieb:
[Description]
Busy([timeout]): This application will indicate the busy condition to
the calling channel. If the optional timeout is specified, the calling
channel
will be hung up after the specified number of seconds. Otherwise, this
application will wait until the calling channel hangs up.
This is what I found when I typed show application busy in the CLI.
Did I interpret it wrong?
regards, Christophorus
Mailinglisten schrieb:
Christophorus Laube schrieb:
hi list,
I set up a new asterisk machine with asterisk 1.2.13 and misdn
0.3.1rc27.
I use an e1 card with sip clients. My extensions look like this:
[E1]
snip...snip
exten = 33006733,1,Set(CALLED=${EXTEN})
exten = 33006733,2,Dial(SIP/[EMAIL PROTECTED])
exten = 33006733-ANSWER,3,Answer()
[SIP]
exten = _X.,1,Noop()
exten = _X.,2,SetCallerPres(allowed_passed_screen)
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40)
exten = _X.-BUSY,4,Busy(1)
But whenever a sip client calls to an exten that is busy through e1 I
get busy tones for 10s before I get disconnected. But I want to have
it only for 1s.
Does anyone know how to fix that?
regards, Christophorus
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AFAIK the BUSY() command has nothing to do with the busy indication.
You can't pass anything to this command.
Check: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Busy
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I think that is something that should be pointed out on the website,
too, then. I did not run that command on the CLI before, sorry.
Is there any output on the CLI that proves the BUSY command is run at
all? Because I don't really know if
exten = _X.-BUSY,4,Busy(1)
is gonna work. I would say something like:
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40,j)
exten = _X.,n+101,Busy(1)
should work if setting the timeout really works that way. Note that the
Dial command has the switch j set which will go to priority n+101 if the
channel is busy.
- Fabian Foerster
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Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Hi Pavel, Thanks for the config! I modified mine so it was more minimal like yours, and it registers just fine now. So much nicer without those big red X's! MG On 13/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: Matt Gibson wrote: Hi Pavel, I tried to implicitly set qualify=no for the sip user, but am still seeing the registering icon for like 10 minutes on the screen of the 7970. It is actually registering, just the phone doesn't think it is. The phones always stay with a little red X on them showing the phone doesn't think it's registered. Weird. maybe some missing in your xml config file? here is my minimalistic .cnf.xml, that works for my 7961 device deviceProtocolSIP/deviceProtocol sshUserIdadmin/sshUserId sshPassword***/sshPassword devicePool dateTimeSetting dateTemplateD-M-Y/dateTemplate timeZoneCentral Europe Standard/Daylight Time/timeZone ntps ntp namentp.ujf.cas.cz/name /ntp /ntps /dateTimeSetting callManagerGroup members member priority=0 callManager ports ethernetPhonePort2000/ethernetPhonePort sipPort5060/sipPort securedSipPort5061/securedSipPort /ports processNodeName192.168.0.100/processNodeName /callManager /member /members /callManagerGroup /devicePool sipProfile sipProxies registerWithProxytrue/registerWithProxy /sipProxies enableVadfalse/enableVad preferredCodecg711a/preferredCodec natEnabled0/natEnabled phoneLabelAsterisk/phoneLabel sipLines line button=1 featureID9/featureID featureLabelSIP 961/featureLabel proxy192.168.0.100/proxy name961/name displayNamePJ7961/displayName authName961/authName authPassword***/authPassword messagesNumber8299/messagesNumber /line line button=2 featureID21/featureID featureLabelEcho test/featureLabel speedDialNumber959/speedDialNumber /line /sipLines dialTemplateDRdialplan.xml/dialTemplate /sipProfile commonProfile phonePassword***/phonePassword /commonProfile loadInformationSIP41.8-2-1S/loadInformation versionStamp1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37/versionStamp /device Thanks for the update! Hopefully these kick ass phones will work better soon! Matt G On 12/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: I'm using 8.2.1 in 7961, it working fine, registration is OK, except I must disable qualify in asterisk (phone doesn't respond to qualify pings), one anoying bug removed is not displaying IP address of sip server (asterisk) in caller id, also same issue with needing rename jar*.sbn file on tftp server anybody made BLF working on 7961 (7970)? PJ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long busy()
Am Mittwoch, den 13.12.2006, 11:47 +0100 schrieb Fabian Foerster:
Is there any output on the CLI that proves the BUSY command is run at
all? Because I don't really know if
exten = _X.-BUSY,4,Busy(1)
is gonna work. I would say something like:
exten = _X.,3,Dial(mISDN/g:E1/${EXTEN},40,j)
exten = _X.,n+101,Busy(1)
should work if setting the timeout really works that way. Note that the
Dial command has the switch j set which will go to priority n+101 if the
channel is busy.
And then, there is ${DIALSTATUS} (without the j switch)...
for me, calling the internal SIP devices looks like
(lots of lines to read database, set variables)
exten = _2XX,116,Dial(${SIPDEVICE},${WAITTIME},gro)
exten = _2XX,117,GotoIf($[X${DIALSTATUS} = XANSWER]?120)
(...)
So if the call was answered, no voicebox will take over... and other
niceties like server-side last-caller-listings etc. In voip-info.org
wiki, there should be plenty information how to use the DIALSTATUS.
I decided to not fully follow the example,but itwas helpful to read
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable
+DIALSTATUS
BR
Anselm
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[asterisk-users] How to temporarily unload modules.
Hi, In what Asterisk file can I edit for me to temporarily unload such modules. But later I woudl still be able to load them. Thanks Angel Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches
Hi, You may want to visit www.procurve.com and look for thier training section there are lots of training materials that can be downloaded. Prices are also posted in this website. Actually, all networking manufacturers has thier training docs posted in their websites. www.3com.com www.nortel.com www.cisco.com - Original Message From: Zeeshan Zakaria [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, December 11, 2006 10:53:53 PM Subject: Re: [asterisk-users] Re: Recommendations for QoS, PoE Switches What's the price for these HP switches? And also I someone can give me a link to some document where I can read about Layer 2 and Layer 3, how they help in VoIP traffic, it'll be helpful. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Have a burning question? Go to www.Answers.yahoo.com and get answers from real people who know.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi Operator
Hi, Actually on my setup all outgoing calls are going trhu a SIP unique account A have a second SIP account with another operator and I would like my setup to use alternatively each of the two accoutns Call 1= Dial SIP/phone1 Call 2= Dial SIP/phone2 Call 3= Dial SIP/phone1 ... If you have an sample please let me know ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi Operator
Hi, Actually on my setup all outgoing calls are going trhu a SIP unique account A have a second SIP account with another operator and I would like my setup to use alternatively each of the two accoutns Call 1= Dial SIP/phone1 Call 2= Dial SIP/phone2 Call 3= Dial SIP/phone1 ... If you have an sample please let me know ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to temporarily unload modules.
/etc/asterisk/modules.conf On 12/13/06, Angel Heart [EMAIL PROTECTED] wrote: Hi, In what Asterisk file can I edit for me to temporarily unload such modules. But later I woudl still be able to load them. Thanks Angel -- Cheap Talk? Check outhttp://us.rd.yahoo.com/mail_us/taglines/postman8/*http://us.rd.yahoo.com/evt=39663/*http://voice.yahoo.comYahoo! Messenger's low PC-to-Phone call rates. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] send fax by Iaxmodem ?
Hi i use now iaxmodem for receive fax and that's work very good with Hylafax ;=) Do you know if we can sent fax using iaxmodem and Hylafax ? when i test: déc 13 13:47:21.12: [13725]: SESSION BEGIN 00014 330426690268 déc 13 13:47:21.12: [13725]: HylaFAX (tm) Version 4.3.0 déc 13 13:47:21.12: [13725]: SEND FAX: JOB 2 DEST 0426690268 COMMID 00014 DEVICE '/dev/iaxmodem1' FROM 'localtest' USER test déc 13 13:47:21.12: [13725]: STATE CHANGE: RUNNING - SENDING déc 13 13:47:21.12: [13725]: -- [12:AT+FCLASS=r] déc 13 13:47:21.12: [13725]: -- [2:OK] déc 13 13:47:21.12: [13725]: MODEM set XON/XOFF/FLUSH: input ignored, output disabled déc 13 13:47:21.12: [13725]: DIAL 0426690268 déc 13 13:47:21.12: [13725]: -- [15:ATDT0426690268\r] déc 13 13:47:21.12: [13725]: -- [11:NO DIALTONE] déc 13 13:47:21.12: [13725]: SEND FAILED: JOB 2 DEST 0426690268 ERR No local dialtone déc 13 13:47:21.12: [13725]: -- [5:ATH0\r] déc 13 13:47:21.12: [13725]: -- [2:OK] déc 13 13:47:21.12: [13725]: MODEM set DTR OFF déc 13 13:47:21.12: [13725]: MODEM set baud rate: 0 baud (flow control unchanged) déc 13 13:47:21.12: [13725]: STATE CHANGE: SENDING - MODEMWAIT (timeout 5) déc 13 13:47:21.12: [13725]: SESSION END Thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to temporarily unload modules.
On Wed, Dec 13, 2006 at 02:03:09AM -0800, Angel Heart wrote: Hi, In what Asterisk file can I edit for me to temporarily unload such modules. But later I woudl still be able to load them. Works fine as long as the module is not in use. asterisk -rx 'unload app_test.so' Later on: asterisk -rx 'load app_test.so' -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re:Re: outgoing call on ISDN PRI
Sorry, sorry !!! I was mixed with another config when I wrote my first
email !!
In fact, User A is registered on Asterisk and user B has a public phone
number (no link with Asterisk).
Our test is : User A calls asterisk server via SIP. As User A context
has a DIAL('user B phone number'),
Asterisk calls user B via ISDN line. Then, user B phone rings and we
can see the caller phone number on
user B phone screen. This caller number is our ISDN line number. What we
would like to do is to hide the caller number (our ISDN line number).
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but it doesn't work.
Do you or anyone know how to hide it?
Thanks you!
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Message: 4
Date: Tue, 12 Dec 2006 19:04:44 +
From: Tim Panton [EMAIL PROTECTED]
Subject: Re: [asterisk-users] outgoing call on ISDN PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
On 12 Dec 2006, at 15:11, Michel wrote:
HEllo list !
When user A calls user B via Asterisk (Users A and B are registered
on the same Asterisk server ) and an ISDN PRI, user B phone
always shows Asterisk server telephone number. How to hide it and
how to forward user A number ?
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but we always see Asterisk server
telephone number !
I'm not getting a clear picture of how the ISDN PRI gets into it if
both users are registered (SIP I assume)
to the same asterisk.
If the call actually goes out via a Public ISDN line, you have to get
the provider to agree to let
you set the outgoing number. Normally they will only let you set it
to one of the inbound numbers
that you have bought from them :-)
If that doesn't help,
please re-phrase the question...
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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Re: [asterisk-users] MeetMe Conferencing and Marked Mode
On 12/13/06, Savoy, Kevin - Williston, ND [EMAIL PROTECTED] wrote: I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks I could be wrong but I reckon one way would be to give the host the admin password. You may or may not need to then add in your DialPlan the logic to mark the user entering the admin password as opposed to users who enter the general PIN. I'm assuming that since meetme is capable to authenticating against 2 PINs, it may auto-mark the user entering the password defined as the admin password in meetme.conf. HTH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] send fax by Iaxmodem ?
Hi Guys, I'm using Asterisk with Hylafax to send and receive faxes, currently only receinving with success. When sending i get this: Dec 13 11:28:07.51: [ 9242]: SESSION BEGIN 00157 03510212079 Dec 13 11:28:07.51: [ 9242]: HylaFAX (tm) Version 4.3.1 Dec 13 11:28:07.51: [ 9242]: SEND FAX: JOB 1 DEST 2079^M COMMID 00157 DEVICE '/dev/ttyIAX' FROM 'Marco Mouta [EMAIL PROTECTED]' USER root Dec 13 11:28:07.51: [ 9242]: STATE CHANGE: RUNNING - SENDING Dec 13 11:28:07.51: [ 9242]: -- [12:AT+FCLASS=1\r] Dec 13 11:28:07.51: [ 9242]: -- [2:OK] Dec 13 11:28:07.51: [ 9242]: MODEM set XON/XOFF/FLUSH: input ignored, output disabled Dec 13 11:28:07.51: [ 9242]: DIAL 2079^M Dec 13 11:28:07.51: [ 9242]: -- [9:ATDT2079\r] Dec 13 11:28:16.70: [ 9242]: -- [4:BUSY] Dec 13 11:28:46.70: [ 9242]: MODEM TIMEOUT: reading line from modem Dec 13 11:28:46.71: [ 9242]: MODEM Timeout Dec 13 11:28:46.71: [ 9242]: SEND FAILED: JOB 1 DEST 2079^M ERR Unknown problem Dec 13 11:28:46.71: [ 9242]: -- [5:ATH0\r] Dec 13 11:28:46.71: [ 9242]: -- [2:OK] Dec 13 11:28:46.71: [ 9242]: MODEM set DTR OFF Dec 13 11:28:46.71: [ 9242]: MODEM set baud rate: 0 baud (flow control unchanged) Dec 13 11:28:46.71: [ 9242]: STATE CHANGE: SENDING - MODEMWAIT (timeout 5) Dec 13 11:28:46.71: [ 9242]: SESSION END I Must say on the reception side is a normal fax connected to pstn line, and to send fax via Asterisk+Hylafax i've tested TE110P and X100P board. I got few sucess with x100p and couldn't send even one with TE110p Any tips? On 12/13/06, Noc Phibee [EMAIL PROTECTED] wrote: Hi i use now iaxmodem for receive fax and that's work very good with Hylafax ;=) Do you know if we can sent fax using iaxmodem and Hylafax ? when i test: déc 13 13:47:21.12: [13725]: SESSION BEGIN 00014 330426690268 déc 13 13:47:21.12: [13725]: HylaFAX (tm) Version 4.3.0 déc 13 13:47:21.12: [13725]: SEND FAX: JOB 2 DEST 0426690268 COMMID 00014 DEVICE '/dev/iaxmodem1' FROM 'localtest' USER test déc 13 13:47:21.12: [13725]: STATE CHANGE: RUNNING - SENDING déc 13 13:47:21.12: [13725]: -- [12:AT+FCLASS=r] déc 13 13:47:21.12: [13725]: -- [2:OK] déc 13 13:47:21.12: [13725]: MODEM set XON/XOFF/FLUSH: input ignored, output disabled déc 13 13:47:21.12: [13725]: DIAL 0426690268 déc 13 13:47:21.12: [13725]: -- [15:ATDT0426690268\r] déc 13 13:47:21.12: [13725]: -- [11:NO DIALTONE] déc 13 13:47:21.12: [13725]: SEND FAILED: JOB 2 DEST 0426690268 ERR No local dialtone déc 13 13:47:21.12: [13725]: -- [5:ATH0\r] déc 13 13:47:21.12: [13725]: -- [2:OK] déc 13 13:47:21.12: [13725]: MODEM set DTR OFF déc 13 13:47:21.12: [13725]: MODEM set baud rate: 0 baud (flow control unchanged) déc 13 13:47:21.12: [13725]: STATE CHANGE: SENDING - MODEMWAIT (timeout 5) déc 13 13:47:21.12: [13725]: SESSION END Thanks bye ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe Conferencing and Marked Mode
Am Dienstag, den 12.12.2006, 13:08 -0600 schrieb Savoy, Kevin - Williston, ND: I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? I understood the docs as the A labeled users (entering through a MeetMe(A) command would be marked, while the w users (MeetMe(...w)) would wait until an A user arrived. Might be wrong though - I don't currently do conferencing because of lack of a zaptel device in my Asterisk box (and kernel is non-modular, and cannot be changed at the moment). BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re:Re: outgoing call on ISDN PRI
Talk to your carrier. Most likely you won't be able to hide it. You
might be able to set it.
Bill
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michel
Sent: Wednesday, December 13, 2006 7:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re:Re: outgoing call on ISDN PRI
Sorry, sorry !!! I was mixed with another config when I wrote my first
email !!
In fact, User A is registered on Asterisk and user B has a public phone
number (no link with Asterisk).
Our test is : User A calls asterisk server via SIP. As User A context
has a DIAL('user B phone number'),
Asterisk calls user B via ISDN line. Then, user B phone rings and we
can see the caller phone number on
user B phone screen. This caller number is our ISDN line number. What we
would like to do is to hide the caller number (our ISDN line number).
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but it doesn't work.
Do you or anyone know how to hide it?
Thanks you!
--
Message: 4
Date: Tue, 12 Dec 2006 19:04:44 +
From: Tim Panton [EMAIL PROTECTED]
Subject: Re: [asterisk-users] outgoing call on ISDN PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
On 12 Dec 2006, at 15:11, Michel wrote:
HEllo list !
When user A calls user B via Asterisk (Users A and B are registered
on the same Asterisk server ) and an ISDN PRI, user B phone
always shows Asterisk server telephone number. How to hide it and
how to forward user A number ?
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but we always see Asterisk server
telephone number !
I'm not getting a clear picture of how the ISDN PRI gets into it if
both users are registered (SIP I assume)
to the same asterisk.
If the call actually goes out via a Public ISDN line, you have to get
the provider to agree to let
you set the outgoing number. Normally they will only let you set it
to one of the inbound numbers
that you have bought from them :-)
If that doesn't help,
please re-phrase the question...
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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Re: [asterisk-users] Re:Re: outgoing call on ISDN PRI
Michel wrote:
Sorry, sorry !!! I was mixed with another config when I wrote my
first email !!
In fact, User A is registered on Asterisk and user B has a public
phone number (no link with Asterisk).
Our test is : User A calls asterisk server via SIP. As User A context
has a DIAL('user B phone number'),
Asterisk calls user B via ISDN line. Then, user B phone rings and we
can see the caller phone number on
user B phone screen. This caller number is our ISDN line number. What
we would like to do is to hide the caller number (our ISDN line number).
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but it doesn't work.
Do you or anyone know how to hide it?
Thanks you!
--
Message: 4
Date: Tue, 12 Dec 2006 19:04:44 +
From: Tim Panton [EMAIL PROTECTED]
Subject: Re: [asterisk-users] outgoing call on ISDN PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
On 12 Dec 2006, at 15:11, Michel wrote:
HEllo list !
When user A calls user B via Asterisk (Users A and B are registered
on the same Asterisk server ) and an ISDN PRI, user B phone
always shows Asterisk server telephone number. How to hide it and
how to forward user A number ?
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but we always see Asterisk server
telephone number !
I'm not getting a clear picture of how the ISDN PRI gets into it if
both users are registered (SIP I assume)
to the same asterisk.
If the call actually goes out via a Public ISDN line, you have to
get the provider to agree to let
you set the outgoing number. Normally they will only let you set it
to one of the inbound numbers
that you have bought from them :-)
If that doesn't help,
please re-phrase the question...
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
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I think your telco is adding the main number assigned to you if you
try not to send one at all or if you send something as a caller ID the
telco didn't allow you to send. IMHO the only thing you can do is ask
the telco not to present the caller ID to the other end. I'm pretty sure
that there is an option to do so in Asterisk, but of course your telco
must support that. Here in Germany this is not a standard feature.
- Fabian Foerster
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[asterisk-users] IAX trunk problem
I wonder if anyone can help me with this. I have 4 sites running asterisk and calls coming into any of these sites are received locally and forwarded to a central operator. E.g. Call comes in on site A and is forwarded to the operator on site B. 99/100 the operator will send the call back to the site from where it came but site B's Asterisk server seems to be staying in the loop. E.g. A B A. I've had a look and can't see anything obvious as I had assumed that asterisk would pass the call off. Thanks Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help getting started with asterisk
On Wed, 2006-12-13 at 15:53 +1100, Paul Hales wrote: What does zttool show? And after you 'modprobe wctdm' what does your dmesg read? /var/log/messages? You should see something about a card being recognised PaulH After I modprobe wctdm, nothing new shows up in /var/log/messages and dmesg is just notices about my firewall. zttool doesn't show much of anything... :( ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re:Re: outgoing call on ISDN PRI
I forgot to mention that the feature in question is called CLIR, or Calling Line Identification Restriction. With that, you can always hide the presentation of your caller ID or do that on a per-call basis. You might want to ask your telco about that. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stress test
Hello peoples, I need to do a test of urgent stress. It know as much as connections simultaneous my equipment is going to do passing codec g729 and g723. Someone knows say me as obtain does him? Andre Luiz Martins mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] MeetMe Conferencing and Marked Mode
I'll give this a try but seems silly to require 2 different extensions for one conference room. Thanks for the input. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Wednesday, December 13, 2006 7:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Conferencing and Marked Mode Am Dienstag, den 12.12.2006, 13:08 -0600 schrieb Savoy, Kevin - Williston, ND: I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? I understood the docs as the A labeled users (entering through a MeetMe(A) command would be marked, while the w users (MeetMe(...w)) would wait until an A user arrived. Might be wrong though - I don't currently do conferencing because of lack of a zaptel device in my Asterisk box (and kernel is non-modular, and cannot be changed at the moment). BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] MeetMe Conferencing and Marked Mode
I did try this and it doesn't work. When logging in with the admin password it still waits for the marked user. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of RR Sent: Wednesday, December 13, 2006 7:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Conferencing and Marked Mode On 12/13/06, Savoy, Kevin - Williston, ND [EMAIL PROTECTED] wrote: I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks I could be wrong but I reckon one way would be to give the host the admin password. You may or may not need to then add in your DialPlan the logic to mark the user entering the admin password as opposed to users who enter the general PIN. I'm assuming that since meetme is capable to authenticating against 2 PINs, it may auto-mark the user entering the password defined as the admin password in meetme.conf. HTH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about hardware
IF I wanted to do the whole sophisticated telephony VoIP stuff asterisk, what hardware would I need? I have a feeling that my fax modem is probably not going to work out. My wife and I have an income of $650 a month. After the first-of-the-month bills are payed, we're lucky if we have $100 left for food and gasoline. I need a solution that's as economical as possible. What exactly do I need in terms of hardware (preferrably specific as in brand names and model numbers)? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [asterisk-users] MeetMe Conferencing and Marked Mode
I was able to get it to work with 2 extensions. One for the host and one for the participants Not the best way to set it up but it works. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Savoy, Kevin - Williston, ND Sent: Wednesday, December 13, 2006 8:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] MeetMe Conferencing and Marked Mode I'll give this a try but seems silly to require 2 different extensions for one conference room. Thanks for the input. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin Hoffmeister Sent: Wednesday, December 13, 2006 7:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe Conferencing and Marked Mode Am Dienstag, den 12.12.2006, 13:08 -0600 schrieb Savoy, Kevin - Williston, ND: I am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? I understood the docs as the A labeled users (entering through a MeetMe(A) command would be marked, while the w users (MeetMe(...w)) would wait until an A user arrived. Might be wrong though - I don't currently do conferencing because of lack of a zaptel device in my Asterisk box (and kernel is non-modular, and cannot be changed at the moment). BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about hardware
cheapy PC (throw away PII is fine) and if you want to use the PSTN, a X100P FXO card. These can be had on ebay for 11 bucks, but I understand that even that pushes the bank some days. You don't need the card, you only need it if you want to receive or place calls on the PSTN. You can use asterisk to do all sorts of ip telephony with just the box that it runs on. Also, be sure to grab the free Oreily book, Asterisk the Future of Telephony. A link to it was posted recently. Michael Sullivan wrote: IF I wanted to do the whole sophisticated telephony VoIP stuff asterisk, what hardware would I need? I have a feeling that my fax modem is probably not going to work out. My wife and I have an income of $650 a month. After the first-of-the-month bills are payed, we're lucky if we have $100 left for food and gasoline. I need a solution that's as economical as possible. What exactly do I need in terms of hardware (preferrably specific as in brand names and model numbers)? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason The place where you made your stand never mattered, only that you were there... and still on your feet ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about hardware
On Wed, 2006-12-13 at 08:29 -0600, jason wrote: cheapy PC (throw away PII is fine) and if you want to use the PSTN, a X100P FXO card. These can be had on ebay for 11 bucks, but I understand that even that pushes the bank some days. You don't need the card, you only need it if you want to receive or place calls on the PSTN. You can use asterisk to do all sorts of ip telephony with just the box that it runs on. Also, be sure to grab the free Oreily book, Asterisk the Future of Telephony. A link to it was posted recently. I ordered the card off ebay. Is there anything else I'd need - special cords, phones, etc? I'd have to try for them next month or after, but I'd prefer to know what they are now so that I can be looking for them... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] anyone used vitelity?
Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about hardware
Don't forget that IF you have NO card, you need to roll ZTDUMMY into the compile. With no card though, you will not be able to read the incoming CLID Also, IF you ever want to progress beyond the X100 card, The Digium cards ( beyond your present budget ( are really intolerant of older PCI buses. Sangoma works with MANY more motherboards. John Novack jason wrote: cheapy PC (throw away PII is fine) and if you want to use the PSTN, a X100P FXO card. These can be had on ebay for 11 bucks, but I understand that even that pushes the bank some days. You don't need the card, you only need it if you want to receive or place calls on the PSTN. You can use asterisk to do all sorts of ip telephony with just the box that it runs on. Also, be sure to grab the free Oreily book, Asterisk the Future of Telephony. A link to it was posted recently. Michael Sullivan wrote: IF I wanted to do the whole sophisticated telephony VoIP stuff asterisk, what hardware would I need? I have a feeling that my fax modem is probably not going to work out. My wife and I have an income of $650 a month. After the first-of-the-month bills are payed, we're lucky if we have $100 left for food and gasoline. I need a solution that's as economical as possible. What exactly do I need in terms of hardware (preferrably specific as in brand names and model numbers)? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallerID Issue (asterisk newbie)
Hi guys. This is my 1st post here (after much reading). I have a test asterisk system setup using X-Lite Soft Phones, and the issue I am running into is that caller id shows up as asterisk on all incoming calls and on all local to local calls (internal). I have showcallerid, etc. configured in zapata.conf, but I'm drawing a blank. When I check my voicemails it tells me that the message is from an unknown caller. I would appreciate any info. -Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audiocodes MediaPack MP-118
Anyone have any experience with the Audiocodes MediaPack MP-118? We are looking at options for a location that wishes to maintain 6 - 8 existing analog phones in addition to a number of new Polycom phones. Thanks, Mike Clark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about hardware
nope, just a regular old phone cord. with that card and a PC, you can receive calls, dial out, terminate SIP, IAX, create an answering machine, run voicemail, talk to jabber servers, all kinds of fun stuff! Asterisk is almost as good as Legos and a lot easier on bare feet at 2am! Michael Sullivan wrote: On Wed, 2006-12-13 at 08:29 -0600, jason wrote: cheapy PC (throw away PII is fine) and if you want to use the PSTN, a X100P FXO card. These can be had on ebay for 11 bucks, but I understand that even that pushes the bank some days. You don't need the card, you only need it if you want to receive or place calls on the PSTN. You can use asterisk to do all sorts of ip telephony with just the box that it runs on. Also, be sure to grab the free Oreily book, Asterisk the Future of Telephony. A link to it was posted recently. I ordered the card off ebay. Is there anything else I'd need - special cords, phones, etc? I'd have to try for them next month or after, but I'd prefer to know what they are now so that I can be looking for them... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason The place where you made your stand never mattered, only that you were there... and still on your feet ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about hardware
You can start off using Soft Phones on your PC (they are free) at 1st once your happy that you want to play voip then you can get either a VOIP hard phone or a VOIP to analog adaptor (Analog Telephone Adaptor), the latter provides you with an FXS port that you can plug a normal phone into or cordless just like the one that your phone company give you. For FXS I have used Linksys SPA3000 which also has the advantage of giving you an FXO port (a connection to the PSTN) as well so you do not need the X100P card. One very nice feature of the SPA3000 is that if the power goes off or your Asterisk box dies the ATA will just bridge the FXO and FXS ports together so that your phone still works, this is an extreamly useful feature when trying to pass the wife test (this is the hardest part of VOIP by a very long chalk). Good Luck Harvey - Original Message - From: Michael Sullivan [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 13, 2006 2:51 PM Subject: Re: [asterisk-users] Question about hardware On Wed, 2006-12-13 at 08:29 -0600, jason wrote: cheapy PC (throw away PII is fine) and if you want to use the PSTN, a X100P FXO card. These can be had on ebay for 11 bucks, but I understand that even that pushes the bank some days. You don't need the card, you only need it if you want to receive or place calls on the PSTN. You can use asterisk to do all sorts of ip telephony with just the box that it runs on. Also, be sure to grab the free Oreily book, Asterisk the Future of Telephony. A link to it was posted recently. I ordered the card off ebay. Is there anything else I'd need - special cords, phones, etc? I'd have to try for them next month or after, but I'd prefer to know what they are now so that I can be looking for them... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by ESVA, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor and Queues
Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my queues set up, but I'm not sure how to combine the two. Ideally, I'd like to only record once the call comes out of queue (no point in recording hold music, unless I want to hear people mumble about how lousy a company we are for placing them on hold ;) ) On a semi-related note, is it possible to determine the extension that pull the call out of queue before the call is bridged? The reason I ask is that I'd like to put the receiving extension in the name of the file that MixMonitor creates. If not, no biggie. Recap: Two queues. First rings for 15 seconds then drops into the second. Second plays music on hold till the call is answered. I want to record the call when it's pulled out of either queue using MixMonitor. Bonus points if I can determine the answering extension before MixMonitor starts (if possible). Any help would be greatly appreciated. Thanks, Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] anyone used vitelity?
We're using vitelity, not in large scale call center type numbers, but any long distance numbers we dial go out their system. They've been working great, but if you expect support for an asterisk system, don't bother calling them. The furthest they'll go is telling you that there are configs on the web and if you're not using a regular IP phone, they can't help you. We did have a hiccup with them yesterday, but other than that, calls are clear and seem to succeed well. On Wed, 2006-12-13 at 08:54 -0600, Curt Shaffer wrote: Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about hardware
The card will let you interface with a regular telephone line instead of VoIP. If you want to use a regular phone instead of the computer softphones, look into the Grandstream handytone devices - they'll make it so your regular telephones can talk to Asterisk. You can make the system work fine with softphones so there's no additional cost at this point... Todd I ordered the card off ebay. Is there anything else I'd need - special cords, phones, etc? I'd have to try for them next month or after, but I'd prefer to know what they are now so that I can be looking for them... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom MyStat
Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Annoying echo echo problem problem ...
Hi Gordon I too have this problem with one of my two BT lines, very very annoying. I was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM), I think the sangoma is very slightly better (less echo) but I might just be kidding myself. Another think I've found it that using a faster CPU (now on a 450 P3 MMX) and compiling Zaptel for mmx helps a little too. The other problem I have is reliable CLID, sangoma and Myphonecall have been very helpful but still no 100% fix :( - Original Message - From: Gordon Henderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 10, 2006 9:24 AM Subject: [asterisk-users] Annoying echo echo problem problem ... I've installed several small asterisk systems now, all mostly the same, a few POTS lines on the outside and a separate analogue phone on the inside (usually a DECT phone of some sorts) and IP phones on the inside, all ticking along quite nicely. However I've hit an issue with the latest one regarding echo )-: Hardware is a TDM400P card. Phone lines are BT, but the telco is Telstra. (there are 2 lines) Asterisk is 1.2.13, Zaptel is 1.2.11. (Kernel 2.6.18 and it's Debian stable, but I doubt that's an issue here) The TDM has all the right UK drivers loaded. Zap/4 and Zap/3 go to the wall sockets, Zap/2 goes to a local analog - GSM gateway box. Zap/2 works perfectly. Zap/3 works perfectly. Zap/4 gives me about half a seconds worth of full-volume echo. Zap/4 gives me about half a seconds worth of full-volume echo. I can say Hello there into a quiet line and hear it echoed back to me in full. I've read and read and re-read just about everything there is to echo and tuning it. Re-compiled the zaptel drivers with the MG2 canceller. I've tried the latest fxotune - which has given me an /etc/fxotune.conf of: 2=5,0,0,0,0,0,0,0,0 3=5,255,252,0,2,254,0,255,255 4=9,255,1,4,0,0,1,255,0 (incidentally, I've never used fxotune on any other installation in the past, they've all just worked) and tried fiddling with the gains (I don't know of a UK based 1Khz 0dB source, but generated my own from another * server using miliwat - how this gets mutated over the wires I don't know, but it seems to work OK to me. Setting Tx gain down to -3 stops dialling working and people I call can't hear much until it's up to at least 3. I can't understand how one line is OK and the other isn't, however when I listen to the line, there is a small amount of noise and a regular low-volume tick tick tick sort of sound. The other line is very quiet. (as it should be) I've had the line re-wired back to the master socket in the building too, so there shouldn't be any issues with internal wiring at all. I've swapped lines into the TDM card and the echo moves with the line which should hopefully eliminate any card problems (unless I have 2 faulty modules?) This one's got me stumped! This one's got me stumped! And of-course, the line is OK when you plug an ordinary analogue phone into it... Any clues, hints, etc. would be most welcome! Thanks, Gordon Output after loading modules, ztcfg and fxotune -s: Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXO (UK mode) Module 2: Installed -- AUTO FXO (UK mode) Module 3: Installed -- AUTO FXO (UK mode) Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Registered tone zone 4 (United Kingdom) Registered tone zone 4 (United Kingdom) -- Setting echo registers: -- Set echo registers successfully -- Setting echo registers: -- Set echo registers successfully -- Setting echo registers: -- Set echo registers successfully /etc/zaptel.conf: fxoks=1 fxsks=2 fxsks=3 fxsks=4 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf: [trunkgroups] [channels] usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no callwaiting=no threewaycalling=yes transfer=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes immediate=no faxdetect=no context=internal signalling=fxo_ks sendcalleridafter=2 rxgain=0 txgain=0 mailbox=100 callerid=100 channel = 1 context=incoming signalling=fxs_ks rxgain=0 txgain=0 group=1 callerid=asreceived usecallerid=no channel = 2 context=incoming signalling=fxs_ks rxgain=0 txgain=0 group=1 callerid=asreceived channel = 3 context=incoming signalling=fxs_ks rxgain=8 txgain=4 group=1 callerid=asreceived channel = 4 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Question about hardware
Speaking of the X100P, I am going to setup an asterisk server next week for a friend's business to replace his aging system. He currently has two voice lines and another line for the fax machine. I was looking at the Sangoma A20200D but that's pretty expensive... We're going to use Grandstream GXP's on desks... Do I need hardware echo cancelation (I'm thinking of using a Dell 2.0 GHz machine)? As Asterisk can handle fax, I was going to drop the 2nd voice line, have the phone company roll busy onto the current fax line, and use that as the second voice line. Can I just use two of the X100 cards? Or is that asking for trouble? thanks Todd On Dec 13, 2006, at 9:56 AM, John Novack wrote: Don't forget that IF you have NO card, you need to roll ZTDUMMY into the compile. With no card though, you will not be able to read the incoming CLID Also, IF you ever want to progress beyond the X100 card, The Digium cards ( beyond your present budget ( are really intolerant of older PCI buses. Sangoma works with MANY more motherboards. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] anyone used vitelity?
I have a development box connected to them and place calls on it from time to time and let family members use it. I have never had any problems, but my usage is rather light and outages might not be noticed with the low volume of calling. On 12/13/06, Curt Shaffer [EMAIL PROTECTED] wrote: Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and Queues
You may want to have a look here: http://astrecipes.net/index.php?n=42 Best regards l. On Wed, 13 Dec 2006 16:15:17 +0100, Jay Moore [EMAIL PROTECTED] wrote: Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my queues set up, but I'm not sure how to combine the two. Ideally, I'd like to only record once the call comes out of queue (no point in recording hold music, unless I want to hear people mumble about how lousy a company we are for placing them on hold ;) ) On a semi-related note, is it possible to determine the extension that pull the call out of queue before the call is bridged? The reason I ask is that I'd like to put the receiving extension in the name of the file that MixMonitor creates. If not, no biggie. Recap: Two queues. First rings for 15 seconds then drops into the second. Second plays music on hold till the call is answered. I want to record the call when it's pulled out of either queue using MixMonitor. Bonus points if I can determine the answering extension before MixMonitor starts (if possible). Any help would be greatly appreciated. Thanks, Jay -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] MixMonitor and Queues
In queues.conf you must have the following under the queues you want to record.
monitor-format=wav49 ; you may also use wav or gsm formats
monitor-join=yes; if you have the latest sox installed,
thiswill join the in and out files into one.
In agents.conf
recordagencalls=yes
monitor-join = yes
recordformat=wav49
savecallsin=/var/www/html/calls ;this is the path where call will be
recorded.
That's all
If you want to change the file name place this in your extensions.conf on a
line prior to sending the call to the queue.
exten= 1097,4,Set(MONITOR_FILENAME=QUEUE-${CALLERID}-${TIMESTAMP})
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore
Sent: Wednesday, December 13, 2006 10:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MixMonitor and Queues
Greetings, all.
I would like to record calls that are entered into queues and I'm not
quite sure how to do it. Here's how I'm currently set up:
- Call comes in and is placed into Queue #1 (which rings all phones for
15 sec).
- If call drops out of this queue, it is placed into Queue #2 (which
plays MoH until the call is picked up).
I've tinkered with MixMonitor and I have my queues set up, but I'm not
sure how to combine the two. Ideally, I'd like to only record once the
call comes out of queue (no point in recording hold music, unless I want
to hear people mumble about how lousy a company we are for placing them
on hold ;) )
On a semi-related note, is it possible to determine the extension that
pull the call out of queue before the call is bridged? The reason I ask
is that I'd like to put the receiving extension in the name of the file
that MixMonitor creates. If not, no biggie.
Recap:
Two queues. First rings for 15 seconds then drops into the second.
Second plays music on hold till the call is answered. I want to record
the call when it's pulled out of either queue using MixMonitor. Bonus
points if I can determine the answering extension before MixMonitor
starts (if possible).
Any help would be greatly appreciated.
Thanks,
Jay
___
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___
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RE: [asterisk-users] MixMonitor and Queues
I've been trying to find where to download the Web Vmail application and instructions on how to install it for Asterisk BE. Any ideas? Thanks Ed Nuñez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Moore Sent: Wednesday, December 13, 2006 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] MixMonitor and Queues Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my queues set up, but I'm not sure how to combine the two. Ideally, I'd like to only record once the call comes out of queue (no point in recording hold music, unless I want to hear people mumble about how lousy a company we are for placing them on hold ;) ) On a semi-related note, is it possible to determine the extension that pull the call out of queue before the call is bridged? The reason I ask is that I'd like to put the receiving extension in the name of the file that MixMonitor creates. If not, no biggie. Recap: Two queues. First rings for 15 seconds then drops into the second. Second plays music on hold till the call is answered. I want to record the call when it's pulled out of either queue using MixMonitor. Bonus points if I can determine the answering extension before MixMonitor starts (if possible). Any help would be greatly appreciated. Thanks, Jay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom MyStat
On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? Doug I think that is strictly a Polycom to Polycom thing (Buddywatch). I do not believe it affects Asterisk (i.e. Busy = DND). With that being said, I don't think it works very well even with all Polycom phones. I can change my status to Busy and look at the other Polycom Phones and they still show me as Online. (Yes, I have bw set to 1.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7970 + New Firmware (8.2)
Matt Gibson wrote: Hi Pavel, Thanks for the config! I modified mine so it was more minimal like yours, and it registers just fine now. So much nicer without those big red X's! MG This modified config works sweet!! Any tricks to get the MWI working? Mark ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] anyone used vitelity?
I think the only gotcha on them is the strange convergence of EXGN and Sixtel that resulted in Vitelity. But hey, maybe they combined their strengths. That said, my sixtel experience was lousy, my EXGN experience ok, and so far, I don't have any real complaints with Vitelity. Trouble-tickets are taken care of promptly... or the one trouble ticket I had, at least. Bruce Reeves wrote: I have a development box connected to them and place calls on it from time to time and let family members use it. I have never had any problems, but my usage is rather light and outages might not be noticed with the low volume of calling. On 12/13/06, *Curt Shaffer* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] anyone used vitelity?
We have been testing them for about a month on outbound only. All I have to say is good luck getting setup. It took us several months just to get a test account and now that we want to actually get service we can't get anyone over there to return our e-mails or calls. They are great for calls but their sales and service departments really need some work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Curt Shaffer Sent: Wednesday, December 13, 2006 8:55 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] anyone used vitelity? Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID Issue (asterisk newbie)
Am Mittwoch, den 13.12.2006, 15:03 + schrieb [EMAIL PROTECTED]: Hi guys. This is my 1st post here (after much reading). I have a test asterisk system setup using X-Lite Soft Phones, and the issue I am running into is that caller id shows up as asterisk on all incoming calls and on all local to local calls (internal). I have showcallerid, etc. configured in zapata.conf, but I'm drawing a blank. When I check my voicemails it tells me that the message is from an unknown caller. I would appreciate any info. Zapata.conf is not usually related to sip device callerid, if you have no Zap interface. Try setting the callerid= stuff in sip.conf appropriately. Mine looks like this, for my desktop phone: [sip504] mailbox=01 callerid=504 type=friend username=sip504 secret=YouDontKnowThis context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm so the callerid= line is what you should adapt. BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Annoying echo echo problem problem ...
On Wed, 13 Dec 2006, Wireless wrote: Hi Gordon I too have this problem with one of my two BT lines, very very annoying. I was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM), I think the sangoma is very slightly better (less echo) but I might just be kidding myself. Another think I've found it that using a faster CPU (now on a 450 P3 MMX) and compiling Zaptel for mmx helps a little too. Hm. Interesting. The CPU is a VIA C3/1GHz unit. (fanless, boots off flash, a no moving parts solution) It has limited MMX instructions though, and requires Zaptel to be compiled for an i586. However, I've not noticed any performance issues with it. We've actually reported it to BT as a fault (the line fails a quiet line test anyway - with a background hiss and ticking) So it's still a bit of a mistery! The other problem I have is reliable CLID, sangoma and Myphonecall have been very helpful but still no 100% fix :( I seem to get CID through OK on BT lines. I'm still battling with a Telewest line though, but I've a feeling theyuse Bell signalling. What annoys me is that a bog-standard DECT phone with CID diaply works perfectly on either a BT or Teleworst line, but the * box fails to see anything other than a BT line.. I'm also using a GSM - analog gateway box, which passes CID in Bell format, even though it has a UK phone socket, but * just doesn't see it at all )-: I'll let you know if BT find anything wrong with the line. Thanks, Gordon - Original Message - From: Gordon Henderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, December 10, 2006 9:24 AM Subject: [asterisk-users] Annoying echo echo problem problem ... I've installed several small asterisk systems now, all mostly the same, a few POTS lines on the outside and a separate analogue phone on the inside (usually a DECT phone of some sorts) and IP phones on the inside, all ticking along quite nicely. However I've hit an issue with the latest one regarding echo )-: Hardware is a TDM400P card. Phone lines are BT, but the telco is Telstra. (there are 2 lines) Asterisk is 1.2.13, Zaptel is 1.2.11. (Kernel 2.6.18 and it's Debian stable, but I doubt that's an issue here) The TDM has all the right UK drivers loaded. Zap/4 and Zap/3 go to the wall sockets, Zap/2 goes to a local analog - GSM gateway box. Zap/2 works perfectly. Zap/3 works perfectly. Zap/4 gives me about half a seconds worth of full-volume echo. Zap/4 gives me about half a seconds worth of full-volume echo. I can say Hello there into a quiet line and hear it echoed back to me in full. I've read and read and re-read just about everything there is to echo and tuning it. Re-compiled the zaptel drivers with the MG2 canceller. I've tried the latest fxotune - which has given me an /etc/fxotune.conf of: 2=5,0,0,0,0,0,0,0,0 3=5,255,252,0,2,254,0,255,255 4=9,255,1,4,0,0,1,255,0 (incidentally, I've never used fxotune on any other installation in the past, they've all just worked) and tried fiddling with the gains (I don't know of a UK based 1Khz 0dB source, but generated my own from another * server using miliwat - how this gets mutated over the wires I don't know, but it seems to work OK to me. Setting Tx gain down to -3 stops dialling working and people I call can't hear much until it's up to at least 3. I can't understand how one line is OK and the other isn't, however when I listen to the line, there is a small amount of noise and a regular low-volume tick tick tick sort of sound. The other line is very quiet. (as it should be) I've had the line re-wired back to the master socket in the building too, so there shouldn't be any issues with internal wiring at all. I've swapped lines into the TDM card and the echo moves with the line which should hopefully eliminate any card problems (unless I have 2 faulty modules?) This one's got me stumped! This one's got me stumped! And of-course, the line is OK when you plug an ordinary analogue phone into it... Any clues, hints, etc. would be most welcome! Thanks, Gordon Output after loading modules, ztcfg and fxotune -s: Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXO (UK mode) Module 2: Installed -- AUTO FXO (UK mode) Module 3: Installed -- AUTO FXO (UK mode) Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Registered tone zone 4 (United Kingdom) Registered tone zone 4 (United Kingdom) -- Setting echo registers: -- Set echo registers successfully -- Setting echo registers: -- Set echo registers successfully -- Setting echo registers: -- Set echo registers successfully /etc/zaptel.conf: fxoks=1 fxsks=2 fxsks=3 fxsks=4 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf: [trunkgroups] [channels] usecallerid=yes cidsignalling=v23
Re: [asterisk-users] Question about hardware
For my home phone system I have an old P-II, which is working perfectly fine for last more than a year now. I had a P-III before that, but one day it died. This P-II is still working and we have no problems with our phone system. I even had conference calls on it with 6 simultaneous users. For the analog phones of home, I have Sipura-1001 attached to this server. Works just great. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MFC/R2 on chan_zap
Hello. I'm trying to setup MFC/R2 signaling, but chan_unicall fails at loading with this message: --- Unable to load module chan_unicall.so Loaded /usr/lib/asterisk/modules/chan_unicall.so = (Unified call processing (UniCall)) == Parsing '/etc/asterisk/unicall.conf': Found 061213-075938 ERROR[11454]: chan_unicall.c:3361 mkintf: Unable to open channel 1: Success here = 0, tmp-channel = 0, channel = 1 061213-075938 ERROR[11454]: chan_unicall.c:4081 setup_unicall: Unable to register channel '1-15' 061213-075938 WARNING[11454]: loader.c:414 __load_resource: chan_unicall.so: load_module failed, returning -1 --- I saw there is some MFC/R2 code on chan_zap.c of asterisk version 1.2.13, has anyone tried it? Thanks signature.asc Description: Esta parte del mensaje está firmada digitalmente ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime +Mysql +Failover
Hoping someone out there has run into this or has some ideas for us. We currently have asterisk set up with Realtime (using mysql) for its extensions,sip and voicemail files. The problem we are trying to solve, is one of a failover mechanism. What if our mysql server went down. Can Realtime be set up with a secondary mysql server to get its data from. We can set up mysql to sync with its fellow server, and maybe when it goes down, it couldn't make any changes (write), but either way, you could still get the extension info, etc, so your phones would still survive a mysql outage. Any ideas? Thanks, Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom MyStat
It still has to go through the upstream pbx/proxy. Each phone doesn't know the location, ie ip address, of the other phones. When the state changes, it should send an updated SIP subscription to Asterisk. -Original Message- From: LST [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 13, 2006 9:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom MyStat On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? Doug I think that is strictly a Polycom to Polycom thing (Buddywatch). I do not believe it affects Asterisk (i.e. Busy = DND). With that being said, I don't think it works very well even with all Polycom phones. I can change my status to Busy and look at the other Polycom Phones and they still show me as Online. (Yes, I have bw set to 1.) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pickup application
Does anyone have the pickup application working? I'm attempting to get it so that a particular extension programmed into a phone can be picked up by another phone with that extension programmed with a speed dial with a 'p' in front... basically, if you dial p100 and extension 100 is ringing, it'll pick up that extension, otherwise it dials the number. The problem I'm having is in the fact that my phones register with mac addresses instead of extensions, so I'm unsure as to what to put in the pickup app args. I've tried mac, extension, sip device name, etc... no luck. Anyone have any ideas? -- Aaron Daniel Senior Voice Analyst Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: New Software available on Cisco.com P0S3-08-5-00
Fyi... My apologies if this is a dupe. -Original Message- From: Cisco Technical Support [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 13, 2006 8:52 AM To: Tim Connolly Subject: New Software available on Cisco.com New software images are available on Cisco.com for the product families that you have selected. If you would like to change your subscription, or unsubscribe, please see the bottom of this e-mail for instructions. This message serves only to advise of new patch availability on Cisco.com (http://www.cisco.com). This is not a direct recommendation to apply the described patch(es) to your system. Please use the release notes, readme(s), your sales team , your advanced services team, TAC, and above all your knowledge of your individual installation to decide if the patch is right for you. Newly Released Voice Software New Software at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960 Filename: P0S3-08-5-00.zip Description : SIP Flash Image for 7940/7960 IP Phone v8.5(0) - Non-CallManager Filename: phrn85s.pdf Description : Release Notes for SIP Flash Image for 7940/7960 IP Phone v8.5(0) - Non-Call Manager Membership Maintenance: Please use these instructions to subscribe or unsubscribe from this list: 1. If you wish to subscribe or unsubscribe from all emails sent by Cisco, please visit your profile manager at http://tools.cisco.com/RPF/profile/profile_management.do to change your preferences. 2. If you wish to subscribe or unsubscribe from all/any software alerts and news, please visit http://www.cisco.com/cgi-bin/Software/Newsbuilder/Builder/VOICE.cgi to change your preferences. Cisco respects your privacy and is committed to protect the personal information that you share with us. Please review Cisco's policy statement at http://www.cisco.com/public/privacy.html, which describes how we collect and use your personal information. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom MyStat
On 12/13/06, Douglas Garstang [EMAIL PROTECTED] wrote: Has anyone ever gotten the Polycom Status feature, accessible via the 'MyStat' soft-key to work? When you change the status in this way, the phone does not send any communication to Asterisk, and it seems to have no effect in incoming calls. So... what's it for? It works somewhat with sipX. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remote-Party-ID and CallerID
I have correct Caller-ID information coming in on the 'Remote-Party-ID' header. The From value is being sent in as Unknown. How could I replace the From value , or CALLERID(all) with the correct values that are in Remote-Party-ID? Or is there a way to tell asterisk to read that header? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Diva Server V-BRI-2 and internal numbers
Hi, I have an asterisk with a DIVA Server V-BRI-2 card, connected to a Siemens PABX. From a SIP phone, I can call other internal SIP phones, external numbers (to PSTN), but I can't call internal phones connected to the internal phone network. When I call 107, which is an internal phone, heres the logs from asterisk: -- Executing Dial(SIP/Greg-081f5a10, CAPI/ISDN1/b:107||rtT) in new stack -- Called ISDN1/b:107 -- CAPI/ISDN1/107-1a is proceeding passing it to SIP/Greg-081f5a10 -- CAPI/ISDN1/107-1a is busy == ISDN1: CAPI Hangingup == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup(SIP/Greg-081f5a10, ) in new stack == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on 'SIP/Greg-081f5a10' in macro 'appel_sortant' == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on 'SIP/Greg-081f5a10' BUT! If I call an internal isdn number like 122 which is a fax, the call is answered. How can I call 107 ? Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Core Dump: create_transaction (p=0x0) at pbx_dundi.c:2787
Anyone seen this...? Is it a known issue?
I'd file a bug, but we're on 1.2.9.13, and every time I file a bug and it isn't
against the latest code I get given crap for it. Given that most of the time
you don't know HOW to reproduce a problem on the latest code anyway, not
accepting bugs from older versions does the community no service, because
potential bugs are never accepted for submission.
(gdb) bt full
#0 0xb7da8d3c in mallopt () from /lib/libc.so.6
No symbol table info available.
#1 0xb7da7e3a in malloc () from /lib/libc.so.6
No symbol table info available.
#2 0xb7b30aa1 in create_transaction (p=0x0) at pbx_dundi.c:2787
trans = (struct dundi_transaction *) 0x0
#3 0xb7b3e616 in find_transaction (hdr=0xbe9fda40, sin=0xbe9ffa40) at
pbx_dundi.c:361
trans = (struct dundi_transaction *) 0x0
#4 0xb7b3e0ef in handle_frame (h=0xbe9fda40, sin=0xbe9ffa40,
datalen=-1209714176) at pbx_dundi.c:1944
trans = (struct dundi_transaction *) 0xbe9ffa40
#5 0xb7b3b3ff in socket_read (id=0x81a61e0, fd=18, events=1, cbdata=0x0) at
pbx_dundi.c:2006
sin = {sin_family = 2, sin_port = 43025, sin_addr = {s_addr =
3415129048}, sin_zero = \000\000\000\000\000\000\000}
res = -1209714176
buf =
t¶\000\000\000\000\211\000\000\006\000\016\f¡\222M\023\004\022KûD\020PÜ\226¶
[EMAIL
PROTECTED](Yi\233TÇ\002Â8èÃ\023\231¸_\220k\0350\227QÙT\031è1ï[oþ}ý\232\\Ã\232ô\224Ægì\026ÀÀuy\231¬å¸\017Úzr)¨åëªb\000nËé5Nºaòdü0¥¦\f®R\237}GDáÄ,\201PFèµÅýÑOû\2076ß©ñ æ¨\022\200\021\202ñI%\t|H\232,m\rh}\235¥|[EMAIL
PROTECTED],¤ûcñ\216æì\214ëS\034\232\016\226449y±\031oñ\201ZÆ_«·c...
len = 16
#6 0x080558cd in ast_io_wait (ioc=0x8134128, howlong=-1209714176) at io.c:284
res = 1
x = 0
origcnt = 1
#7 0xb7b35e6f in network_thread (ignore=0x0) at pbx_dundi.c:2106
res = -1209714100
#8 0xb7ef9ed8 in pthread_start_thread () from /lib/libpthread.so.0
No symbol table info available.
#9 0xb7df87ea in clone () from /lib/libc.so.6
No symbol table info available.
(gdb)
Doug.
___
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[asterisk-users] record time with phones option buttons
Anyone able to point me the right direction for the following would be helpful. I have a client that needs to keep detailed time for how long their Customer Service Reps. Spend on different subject with the customers. i.e. All CSR's are trained to take all types of calls. For regulatory reasons they have to keep track of how long they spend talking to a customer about different offerings. A call comes in and they cross sell for another division in the company and if the customer is interested they need to record their time to that division. Say we have Cisco 7940's or 7960's or any phone that has the additional buttons other than call appearance. Can we program those buttons to start recording that reps time to the correct division. i.e. CSR talks call for division #1 they press the first button on the phone. Same CSR cross sells for division #2 they than press second button to record time for that division Throwing all this into a database to pull out realtime, daily or weekly would be perfect. Thanks -Matt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with voicemail
I'm looking to use * for a HQ/branch office topology with fairly few calls over the WAN. The questions I have all pertain to the following architectural pic: http://www.45891.com/misc/arch.jpg I'm looking at a distributed architecture so users are somewhat functional when the link to HQ is down, with a centralized voicemail server to allow for transfer of voicemail messages from user to user, on both the VoIP and legacy system (voicemail being on a dedicated * box). 1. Thanks to jporier who can be found at ccu.edu, I figured out how to deal with MWI for all the remote servers by mounting the voicemail directory via NFS from VMAIL1 onto the VOIPx servers which host the actual phones. Then sticking a msg0.txt file into the directory makes the blinky light go on the phones. So far so good. What I'm asking the list for is either a brief code snippet or pointers to a doc/link on how to setup the following: A. None of the VOIPx servers have vmail enabled on them. When someone gets dumped to voicemail, I envision the call being transferred to the VMAIL1 server and it routing it directly to a mailbox for the user. B. VMAIL1 has no user extensions on it, just mailboxes. It gets a call on the trunk and dumps it to the appropriate vmail box based on the extension that was called. C. How do I force the vmail to go down the trunk to VMAIL1? D. How do I catch it on the other end and stick it only in a mailbox? Basically, how do I split the voicemail transfer off the local box to another? Now for a couple of architectural questions: 1. When a caller rings thru the TANDEM1 box to a VOIP1 extension, and then gets dumped to vmail, does the call go TANDEM1-VOIP1-VMAIL1 or does VOIP1 hand it off so it's only TANDEM1-VMAIL1, presuming all IAX2 trunks are running a matching subset of codecs? 2. Same thing for intracompany calls. If VOIP2 calls VOIP1 user via the tandem and gets dumped to vmail, does it go VOIP2-VOIP1-VMAIL1 or VOIP2-VMAIL1? When user is talking on PSTN over Teliax, I can see TANDEM1 doing the transcoding if necessary and bridging via IAX2 show peers. This leads me to believe it would go the former route, not the latter. If it is the former, is there a way to make it do the latter? 3. For the TANDEM1 to VMAIL1 trunk, does it make sense to do G711 as well on the trunk so it can transfer without transcoding to the voicemail box (user dials the voicemail number DID on PRI from Embarq, hits the mapping on the tandem and goes down the VMAIL1 trunk). 4. Does it make sense to have a redundant tandem running on another box and split the PRI's from the IAX trunks? Embarq is looking into forwarding the PRI DID blocks to the pilot number for the IAX2 trunk from Teliax so when it goes down or is all-trunks-busy, it comes down the 'Net pipe. Nice to have Embarq on one side of the road ariel and TW underground on the other side with separate entrances. 5. When a call is hairpinned in TANDEM1 from the Embarq PRI to the tie PRI's, is there any CPU overhead involved or is it basically done in the card, presuming matching codecs on the PRI's? Card is a digium TE405P quad PRI card. Some implementation notes: 1. All the boxes with IP addresses shown in the pic are setup. I have successful calls going Teliax - Tandem - VOIP1 and also back out to the PSTN via the Tandem. VOIP2 comes up tomorrow. PRI's are a middle of the night job later this week. 2. All are running Trixbox 2.0b2. 3. We're playing with codecs to see what gives the best quality for the bandwidth. Voip-info.org seems to point towards ilbc as having the lowest overhead, followed by gsm and g729. I presume if we want to bring fax in off the Embarq PRI and/or Teliax we're going to have to use G.711u thru to the Hylafax server with iaxmodem. Anybody have any experience with bringing fax in over a IAX2 trunk from Teliax (or any other voip provider for that matter)? We're switching this Thursday to a 10Mbps symmetric fiber connection from Time Warner Business Class. Once I get this working, I'm willing to write up a how-to (I'm going to have to anyways for documentation, just needs to be sanitized) and put a pointer or the doc on voip-info.org Thanks in advance. EKG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MFC/R2 on chan_zap
On Wed, Dec 13, 2006 at 09:12:13AM -0500, Alejandro Rios Peña wrote: Hello. I'm trying to setup MFC/R2 signaling, but chan_unicall fails at loading with this message: --- Unable to load module chan_unicall.so Loaded /usr/lib/asterisk/modules/chan_unicall.so = (Unified call processing (UniCall)) == Parsing '/etc/asterisk/unicall.conf': Found 061213-075938 ERROR[11454]: chan_unicall.c:3361 mkintf: Unable to open channel 1: Success here = 0, tmp-channel = 0, channel = 1 061213-075938 ERROR[11454]: chan_unicall.c:4081 setup_unicall: Unable to register channel '1-15' 061213-075938 WARNING[11454]: loader.c:414 __load_resource: chan_unicall.so: load_module failed, returning -1 --- I saw there is some MFC/R2 code on chan_zap.c of asterisk version 1.2.13, has anyone tried it? It was removed for being non-functional. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SRV Entries
I saw on a mailing list for digium that back in March, they were looking to get SRV working properly. Was this ever repaired? If so, is it just a matter of adding 2 entries to tinydns data file, and then point the res_mysql.conf file to point to the new hostname (astmysql.yournet.com)? Trying any way possibly for redundancy. Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Remember last IP address of IAX client
Hello, does anybody know if it is possible to save the IP address of an IAX client logging into asterisk into the DB for future reference? I.e. one could distinguish between cases, where the client was last seen on the local net or on the road... even when it is not currently online. Cheers, Arik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to define a secure trunk
Hello I would like to define a trunk from my Asterisk to a VoIP provider, but I want to make it secure, because its through the Internet. I want to be sure no one makes calls as being me, and that my calls aren't intercepted. Is it possible to define encrypted trunks? And should I define the trunk in SIP, IAX or something else? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime +Mysql +Failover
On 12/13/06, Rob Schall [EMAIL PROTECTED] wrote: Hoping someone out there has run into this or has some ideas for us. We currently have asterisk set up with Realtime (using mysql) for its extensions,sip and voicemail files. The problem we are trying to solve, is one of a failover mechanism. What if our mysql server went down. Can Realtime be set up with a secondary mysql server to get its data from. We can set up mysql to sync with its fellow server, and maybe when it goes down, it couldn't make any changes (write), but either way, you could still get the extension info, etc, so your phones would still survive a mysql outage. Any ideas? Thanks, Rob I don't think Realtime can be setup with a secondary server (someone please correct me if I'm wrong). Two possibilities come to mind... 1. You can run MySQL in an HA arangement with on box as the hot standby. 2. If you can allow for ocassional asterisk reloads, you could use Realtime Static Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multi Operator
At 04:27 AM 12/13/2006, you wrote:
A have a second SIP account with another operator and I would like my setup
to use alternatively each of the two accoutns
Call 1= Dial SIP/phone1
Call 2= Dial SIP/phone2
Call 3= Dial SIP/phone1
...
If you have an sample please let me know
Something like this should work.
Ira
[GLOBAL]
LINE_CHOICE=1
[out]
exten = s,1, set(LINE_CHOICE=$[${LINE_CHOICE} + 1])
exten = s,n,gotoif($[${LINE_CHOICE} = 2]?continue_here)
exten = s,n,set(LINE_CHOICE=1)
exten = s,n(continue_here),
exten = s,n,dial(SIP/phone${LINE_CHOICE})
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Re: [asterisk-users] SRV Entries
On 12/13/06, Rob Schall [EMAIL PROTECTED] wrote: I saw on a mailing list for digium that back in March, they were looking to get SRV working properly. Was this ever repaired? If so, is it just a matter of adding 2 entries to tinydns data file, and then point the res_mysql.conf file to point to the new hostname (astmysql.yournet.com)? Trying any way possibly for redundancy. Rob Asterisk will do SRV lookups, it just does not fail to the next record if the first is unavailable as SRV was intended. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring VoIP latency and packet loss
iftop On 12/12/06, Mochamad Susantok [EMAIL PROTECTED] wrote: Dear all, Are there anyone have ben to use some tool or method to measure latency and packet loss for VoIP packet ? - This email was sent using Student EEPIS-Webmail. http://student.eepis-its.edu/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID Issue (asterisk newbie)
Anselm Martin Hoffmeister wrote: Am Mittwoch, den 13.12.2006, 15:03 + schrieb [EMAIL PROTECTED]: Hi guys. This is my 1st post here (after much reading). I have a test asterisk system setup using X-Lite Soft Phones, and the issue I am running into is that caller id shows up as asterisk on all incoming calls and on all local to local calls (internal). I have showcallerid, etc. configured in zapata.conf, but I'm drawing a blank. When I check my voicemails it tells me that the message is from an unknown caller. I would appreciate any info. Zapata.conf is not usually related to sip device callerid, if you have no Zap interface. Try setting the callerid= stuff in sip.conf appropriately. Mine looks like this, for my desktop phone: [sip504] mailbox=01 callerid=504 type=friend username=sip504 secret=YouDontKnowThis context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm so the callerid= line is what you should adapt. BR Anselm I've seen this recently when the caller ID comes in NPANXXy somehow the very long callerid isn't handled well. I ended up peeling off the first 10 digits and re-stuffing the callerid with that ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playing a sound file on handset pickup
I've added the ability for a user to record a custom message associated with a special IVR menu for occasions when business will be closed for some non-standard amount of time (Maybe 4 days at Christmas...) They just dial 800, record the message then hang up and dial 801 to enable it. Presumably, when they return after the holiday, they should dial 802 to disable it and return to the normally scheduled menus. But they will most likely forget so I'd like to set up some type of reminder functionality; perhaps playing a message back to them stating that the custom message is still enabled before giving them dialtone or something to the same effect. Is this possible and can anyone offer recommendations? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to define a secure trunk
http://www.voip-info.org/wiki/view/IAX+encryption Joao Pereira wrote: Hello I would like to define a trunk from my Asterisk to a VoIP provider, but I want to make it secure, because its through the Internet. I want to be sure no one makes calls as being me, and that my calls aren't intercepted. Is it possible to define encrypted trunks? And should I define the trunk in SIP, IAX or something else? Thanks Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing a sound file on handset pickup
John French wrote: I've added the ability for a user to record a custom message associated with a special IVR menu for occasions when business will be closed for some non-standard amount of time (Maybe 4 days at Christmas...) They just dial 800, record the message then hang up and dial 801 to enable it. Presumably, when they return after the holiday, they should dial 802 to disable it and return to the normally scheduled menus. But they will most likely forget so I'd like to set up some type of reminder functionality; perhaps playing a message back to them stating that the custom message is still enabled before giving them dialtone or something to the same effect. Is this possible and can anyone offer recommendations? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Why not just add that functionality to the s extension? If no extension is given, they will end up there, won't they? So if that I'm not here message is set up, and the client picks up the phone, we assume that he/she is back and thus delete the notification without notice. - Fabian Foerster ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 realtime with mysql 5.0 and unixODBC.
Hi Lan - I am trying to upgrade my testing asterisk realtime 1.2.13 with MySQL 5.0 and unixODBC to the beta asterisk 1.4. I run the make and make install for the asterisk-addon just fine, It created the modules res_config_mysql.so and cdr_addon_mysql.so without any problem or error. However, when I run the asterisk, it comes up with the error : == Parsing '/etc/asterisk/res_mysql.conf': Found asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: mysql_init One question: Did you remove the old 1.2.x addon modules from your modules directory (/usr/lib/asterisk/modules) before you installed the new 1.4.x addons? - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ParkAndAnnounce + Paging
It is possible to announce the parking position through a paging to a group of extensions? I would like that when someone parks a call, some phones will announce with the speaker the position. Something like: exten = s,1,ParkAndAnnounce(call-parked-at:PARKED|30|PAGE(LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]|) Is there a way, maybe with a different approach? I think your method should work. Have you tried it yet? It's a very good idea, BTW. Talk about an auto-attendant! - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallerID Issue (asterisk newbie)
Hi everybody... I have a similar problem... I don't get the ID of the person that i called on my phone... Does anyone know something about this problem? greets, Sven 2006/12/13, Bruce Ferrell [EMAIL PROTECTED]: Anselm Martin Hoffmeister wrote: Am Mittwoch, den 13.12.2006, 15:03 + schrieb [EMAIL PROTECTED]: Hi guys. This is my 1st post here (after much reading). I have a test asterisk system setup using X-Lite Soft Phones, and the issue I am running into is that caller id shows up as asterisk on all incoming calls and on all local to local calls (internal). I have showcallerid, etc. configured in zapata.conf, but I'm drawing a blank. When I check my voicemails it tells me that the message is from an unknown caller. I would appreciate any info. Zapata.conf is not usually related to sip device callerid, if you have no Zap interface. Try setting the callerid= stuff in sip.conf appropriately. Mine looks like this, for my desktop phone: [sip504] mailbox=01 callerid=504 type=friend username=sip504 secret=YouDontKnowThis context=sipclient host=dynamic nat=yes disallow=all allow=alaw allow=gsm so the callerid= line is what you should adapt. BR Anselm I've seen this recently when the caller ID comes in NPANXXy somehow the very long callerid isn't handled well. I ended up peeling off the first 10 digits and re-stuffing the callerid with that ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Programming soft buttons on the IP601?
When the IP601 is sitting unused, it uses the first 2 of the 4 soft buttons under the screen. The third one is empty, which is good because it is used for Exit. I would like to be able to use that 4th button for group pickup (*8#) and have it read Pickup. Is this possible? If so, how? Thanks, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom IP4000 and vsftpd 2.0.1
Is anyone else having trouble getting a Polycom IP4000 (running SIP 1.6.7 and BootROM 3.1.3) to download its configuration files from a vsftpd 2.0.1 server? We have 100+ IP501s that manage this without problems, but the IP4000 keeps timing out. We have opened a case with Polycom, but they are insisting that it is our configuration files that are at fault, even though the phone times out on bootrom.ld, long before it attempts to load the configuration files. I did turn up some postings about IP501s, BootROM 3.1.3 and vsftpd 2.0.3, and wonder if this might be a similar issue. CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Core Dump: create_transaction (p=0x0) at pbx_dundi.c:2787
In article [EMAIL PROTECTED], Douglas Garstang [EMAIL PROTECTED] wrote: Anyone seen this...? Is it a known issue? I'd file a bug, but we're on 1.2.9.13, and every time I file a bug and it isn't against the latest code I get given crap for it. Given that most of the time you don't know HOW to reproduce a problem on the latest code anyway, not accepting bugs from older versions does the community no service, because potential bugs are never accepted for submission. (gdb) bt full #0 0xb7da8d3c in mallopt () from /lib/libc.so.6 No symbol table info available. #1 0xb7da7e3a in malloc () from /lib/libc.so.6 No symbol table info available. #2 0xb7b30aa1 in create_transaction (p=0x0) at pbx_dundi.c:2787 trans = (struct dundi_transaction *) 0x0 Hmmm, that will be a tricky one to track down. There's no reason to get a core dump from within malloc() unless something else has previously stomped outside of its own malloced area, smashing the free list. So the problem is likely not within create_transaction(), but caused sometime before, possibly in some completely unrelated code. Is it repeatable, or just happens at random (or even just once)? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] webvoicemail
On 12/13/06, Ed Nuñez [EMAIL PROTECTED] wrote: I've been trying to find where to download the Web Vmail application and instructions on how to install it for Asterisk BE. Any ideas? Is this any different than the vmail.cgi that comes with the open version? Otherwise, you will just need to grab a compiled copy off of another box. Only needs vmail.cgi and a couple of supporting graphics. -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Measuring VoIP latency and packet loss
Carlos Rojas wrote: iftop On 12/12/06, *Mochamad Susantok* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Dear all, Are there anyone have ben to use some tool or method to measure latency and packet loss for VoIP packet ? Commercial or Open Source? For Open Source, try IPTraf, PKStat, Netperf, Softflowd, MRTG, make your own is pretty much what I try to do... Commercial? Opera from Opticom, or Fluke Networks' NetTools ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP4000 and vsftpd 2.0.1
Do you have the latest firmware files from polycom and sample configurations? Can you get the phone to accept those? Any reason why you are using FTP? Http has worked without a hitch. What does your logs say? On 12/13/06, Anthony Rodgers [EMAIL PROTECTED] wrote: Is anyone else having trouble getting a Polycom IP4000 (running SIP 1.6.7 and BootROM 3.1.3) to download its configuration files from a vsftpd 2.0.1 server? We have 100+ IP501s that manage this without problems, but the IP4000 keeps timing out. We have opened a case with Polycom, but they are insisting that it is our configuration files that are at fault, even though the phone times out on bootrom.ld, long before it attempts to load the configuration files. I did turn up some postings about IP501s, BootROM 3.1.3 and vsftpd 2.0.3, and wonder if this might be a similar issue. CP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playing a sound file on handset pickup
How about put it in the dial plan? So anytime you try to make an outbound call it would play a reminder saying that the alternate greeting is enabled. You could just use a DB variable. On 12/13/06, Mailinglisten [EMAIL PROTECTED] wrote: John French wrote: I've added the ability for a user to record a custom message associated with a special IVR menu for occasions when business will be closed for some non-standard amount of time (Maybe 4 days at Christmas...) They just dial 800, record the message then hang up and dial 801 to enable it. Presumably, when they return after the holiday, they should dial 802 to disable it and return to the normally scheduled menus. But they will most likely forget so I'd like to set up some type of reminder functionality; perhaps playing a message back to them stating that the custom message is still enabled before giving them dialtone or something to the same effect. Is this possible and can anyone offer recommendations? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Why not just add that functionality to the s extension? If no extension is given, they will end up there, won't they? So if that I'm not here message is set up, and the client picks up the phone, we assume that he/she is back and thus delete the notification without notice. - Fabian Foerster ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Programming soft buttons on the IP601?
Hi Warren - When the IP601 is sitting unused, it uses the first 2 of the 4 soft buttons under the screen. The third one is empty, which is good because it is used for Exit. I would like to be able to use that 4th button for group pickup (*8#) and have it read Pickup. Is this possible? If so, how? In short, no, the soft-buttons cannot be user-programmed. Many on this list have tried to reprogram these buttons, but I'm not aware of anyone who's actually gotten anything to work. I'd love for someone to prove me wrong, of course. The hard buttons are programmable. With the 2.0.x firmwares you can program a string of digits (*8#) by re-programming one of the hard keys as a speed dial. The services button isn't normally used, so you could use it for this purpose. I generally reprogram the Directories and Call Lists keys, too, as they're redundant. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diva Server V-BRI-2 and internal numbers
On Wed, 13 Dec 2006, Gregory Duchatelet wrote: Hi, I have an asterisk with a DIVA Server V-BRI-2 card, connected to a Siemens PABX. From a SIP phone, I can call other internal SIP phones, external numbers (to PSTN), but I can't call internal phones connected to the internal phone network. When I call 107, which is an internal phone, heres the logs from asterisk: -- Executing Dial(SIP/Greg-081f5a10, CAPI/ISDN1/b:107||rtT) in new stack -- Called ISDN1/b:107 -- CAPI/ISDN1/107-1a is proceeding passing it to SIP/Greg-081f5a10 -- CAPI/ISDN1/107-1a is busy == ISDN1: CAPI Hangingup == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup(SIP/Greg-081f5a10, ) in new stack == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on 'SIP/Greg-081f5a10' in macro 'appel_sortant' == Spawn extension (macro-appel_sortant, s, 2) exited non-zero on 'SIP/Greg-081f5a10' BUT! If I call an internal isdn number like 122 which is a fax, the call is answered. How can I call 107 ? It looks like 107 is busy ;-) Please increase verbosity, like set verbose 5 capi debug to see what is happening. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ssh access using zaptel channel to dial in.
Has anyone done this, or have a thought on how to do it. I forsee it working like this... Dial in to a main greeting, dial an extension using a modem string like 782-,,,##409*. The extension would some kind of modem emulator. I know this compromises security. I was hoping to use an authenticate app in there as well. My main question is using the zap hardware and some kind of dialplan app to accomplish this Jordan Novak ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ZAP multiline handset questions
Hi All - I haven't worked much with ZAP handsets before, but I've got a client who is insistent on using a particular phone. My questions: 1. With multiline analog phones, if I've got multiple phones, each connected to a different FXS interface, is there a way to make the line status lights on the other phones show that a particular FXO is in use (like a key system, or like SIP hinting)? 2. Does anyone know of a good analog cordless phone (independent of any base desk phone) that can handle multiple lines? Thanks! Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ssh access using zaptel channel to dial in.
Hi Jordan - Has anyone done this, or have a thought on how to do it. I forsee it working like this... Dial in to a main greeting, dial an extension using a modem string like 782-,,,##409*. The extension would some kind of modem emulator. I know this compromises security. I was hoping to use an authenticate app in there as well. My main question is using the zap hardware and some kind of dialplan app to accomplish this This sounds like a whole lot of unnecessary complication. Why not just use a regular old modem connected to a serial interface on the computer you want to get CLI access to? No need to involve asterisk at all. For security, let the OS handle authentication. Old school? Yes, but it works. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
