[asterisk-users] Problem with pjsip
Hello everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint 'Biv_Sortie' - 'from_user' field contains invalid character '@' [2023-06-08 13:19:03] ERROR[185091]: config_options.c:798 aco_process_var: Error parsing from_user=75b55btqu...@orange-obs.fr at line 0 of == chan_pjsip.so => (PJSIP Channel Driver) 1) Error with "@" character which constitutes URI and authuser see excerpt from pjsip.conf. [transport-udp] type = transport protocol=udp bind=0.0.0.0:5060 local_net=172.16.1.0/255.255.255.0 [reg_orange-obs.fr] type = registration retry_interval = 120 max_retries = 10 expiration = 120 transport = transport-udp outbound_auth = auth_reg_orange-obs.fr client_uri = sip:+3313445x...@orange-obs.fr server_uri = sip:orange-obs.fr [auth_reg_orange-obs.fr] type=auth password=3314C9BA9688C2AA username = 75b55btqu...@orange-obs.fr [Biv_Sortie] type = aor contact = sip:75b55btqu...@orange-obs.fr@orange-obs.fr default_expiration = 3600 [Biv_Sortie] type = identify endpoint = Biv_Sortie match = orange-obs.fr [Biv_Sortie] type=auth username = Biv_Sortie password=3314C9BA9688C2AA [Biv_Sortie] type=endpoint context = Isdn_Inbound dtmf_mode=rfc4733 disallow=all allow = g722, alaw, g729 direct_media=no trust_id_inbound = yes send_rpid=yes from_user = 75b55btqu...@orange-obs.fr from_domain = orange-obs.fr language = en allow_subscribe = yes auth = Biv_Exit outbound_auth = Biv_Sortie aors = Biv_Sortie Question how can I solve this character problem "@"? 2) resolution of the orange-obs.fr DNS. I am attaching an extract from the documentation that Orange issued in 2015 SIP/Internet is described in RFC3261 and following. THE SIP/IMS is described by 3GPP standards. It's not the same SIP. In the Internet world, VoIP machines route SIP messages to the IP addresses of the FQDNs of the SIP URIs (VoIP domain). In the 3GPP world, SIP messages are routed to an I/P-CSCF (depending on whether we are in interco or in IPBX) which has a different FQDN from the VoIP domain. BIV SIP – P-CSCF FQDN: pcscfgm.orange-obs.fr, resolved by DNS voice – VoIP domain: orange-obs.fr, not resolved by voice DNS. ex : INVITE sip:0142277...@orange-obs.fr SIP/2.0 2 The VoIP/Internet machine will not be able to determine the address recipient of SIP messages. run the command “nslookup pcscfgm.orange-obs.fr” and note the returned IP address 217.167.210.X – add this address in the /etc/hosts file of the PBX: 217.167.210.X pcscfgm.orange-obs.fr orange-obs.fr Note that it works with sip.conf . The current installation is operational with the information provided by /etc/hosts below the debug in asterisk 19.6 [2023-06-08 13:37:17] DEBUG[185433]: res_config_odbc.c:115 custom_prepare: Skip: 0; SQL: SELECT * FROM ps_auths WHERE id = ? [2023-06-08 13:37:17] DEBUG[185433]: res_config_odbc.c:134 custom_prepare: Parameter 1 ('id') = 'auth_reg_orange-obs.fr' [2023-06-08 13:37:17] DEBUG[185433]: res_odbc.c:808 ast_odbc_release_obj: Releasing ODBC handle 0x55855d1977d0 into pool [2023-06-08 13:37:17] DEBUG[185433]: config.c:3847 ast_parse_arg: extract uint from [32] in [0, 4294967295] gives [32](0) [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip_outbound_registration.c:699 handle_client_registration: Outbound REGISTER attempt 2 to 'sip:orange-obs.fr' with client 'sip:+3313445x...@orange-obs.fr' [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:475 sip_resolve: Performing SIP DNS resolution of target 'orange-obs.fr' [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:502 sip_resolve: Transport type for target 'orange-obs.fr' is 'UDP transport' [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:545 sip_resolve: [0x55855d769c88] Created resolution tracking for target 'orange-obs.fr' [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:174 sip_resolve_add: [0x55855d769c88] Added target 'orange-obs.fr' with record type '35', transport 'UDP transport', and port '5060' [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:174 sip_resolve_add: [0x55855d769c88] Added target '_sip._udp.orange-obs.fr' with record type '33', transport 'UDP transport', and port '5060' [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:174 sip_resolve_add: [0x55855d769c88] Added target 'orange-obs.fr' with record type '1', transport 'UDP transport', and port '5060' [2023-06-08 13:37:17] DEBUG[185433]: res_pjsip/pjsip_resolver.c:616 sip_resolve: [0x55855d769c88] Starting initial resolution using parallel queries for target 'orange-obs.fr' [2023-06-08 13:37:17] DEBUG[185340]: dns.c:555 ast_search_dns_ex: DNS search failed for orange-obs.fr [2023-06-08 13:37:17] DEBUG[185340]: dns_system_resolver.c:154 dns_system_resolver_process_query: DNS search failed for query: 'orange-obs.fr' [2023-06-08 13:37:17]
Re: [asterisk-users] Problem with AudioCodes MP-114 ATA
hi, quite unlikely (besides of an defect) that the behaviour of your AudioCodes or Asterisk changed "from alone"... something must have changed. What does the logs say (from asterisk... do you see register-events? and from you AudioCodes?) The AudioCodes Devices can export and restore their config... do you have a backup? regards, yves Am 10.01.2019 um 19:51 schrieb Tech Support: All; I have an AudioCodes MP-114 four FXS ATA that recently stopped registering to my PBX. I’m pulling my hair out here trying to figure out the root cause without much success. Does anyone have a sample config file that I could use as a sample? Any insight at all would be greatly appreciated. Thanks Much; John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use AGi Commands without script in Dialplan
Am 08.10.2018 um 13:02 schrieb Antony Stone: On Monday 08 October 2018 at 12:44:43, Yves wrote: I am looking for an easy way to execute any AGI Command directly from the dialplan without the need to call an external script. The whole point of AGI is that it calls an external script in order to replace commands in the dialplan. Executing an AGI command without an external script doesn't make sense. Antony. Hi Antony, thanks for your answer, even if it is a bit disappointing for me. I understand the point... but... why aren´t then all AGI-Commands also available as Dialplan Functions? I can only find a small amount of functions for the dialplan that could be seen as an equivalent or near-equivalent of an AGI Command... thank you, Yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use AGi Commands without script in Dialplan
Am 09.10.2018 um 13:56 schrieb Joshua Colp: On Mon, Oct 8, 2018, at 7:44 AM, Yves wrote: Hello, everybody, often it is necessary to issue a single AGI command... How can I realize this within a normal dialplan processing without having to go the circumstantial way through an AGI script every time? Why is it not possible to use the AGI commands like other functions within the dialplan? Although there are many dialplan functions that can be used as a substitute for one or the other AGi command, or whose results are the same, but not always... Example: AGI_Command "Set Autohangup"... There is no way (at least of what I know) to set this AutoHangup feature for a "normal" Call within the dialplan... and again, this is just an example. I am looking for an easy way to execute any AGI Command directly from the dialplan without the need to call an external script. In particular for this it can done in dialplan using the TIMEOUT dialplan function[1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_TIMEOUT Hi, thank you, great to have a replacement for this particular function. Yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use AGi Commands without script in Dialplan
Hello, everybody, often it is necessary to issue a single AGI command... How can I realize this within a normal dialplan processing without having to go the circumstantial way through an AGI script every time? Why is it not possible to use the AGI commands like other functions within the dialplan? Although there are many dialplan functions that can be used as a substitute for one or the other AGi command, or whose results are the same, but not always... Example: AGI_Command "Set Autohangup"... There is no way (at least of what I know) to set this AutoHangup feature for a "normal" Call within the dialplan... and again, this is just an example. I am looking for an easy way to execute any AGI Command directly from the dialplan without the need to call an external script. Thank you, Yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Astricon is coming up October 9-11! Signup is available at: https://www.asterisk.org/community/astricon-user-conference Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trying to add MoH to conference bridge
could you switch asterisk to verbose >=3 and show the output from the cli? which version of asterisk do you use? yves Am 23.05.2018 um 23:23 schrieb Mike Diehl: Hi all, I've got an AGI script that launches the conference bridge with a line like: "$main::agi->exec(ConfBridge,$conf,default_bridge,default_user,$menu_profile)" The $conf variable contains the room number. I'm trying to configure it so that when only one person is in the conference, they hear moh. My /etc/asterisk/confbridge.conf looks like: === [general] [default_bridge] type=bridge [default_user] type=user quiet=no announce_join_leave=yes music_on_hold_class=default music_on_hold_when_empty=yes [default_menu] type=menu 0=playback_and_continue(/none) 1=increase_listening_volume 2=toggle_mute 3=increase_talking_volume 4=reset_listening_volume 5=admin_toggle_mute_participants 6=reset_talking_volume 7=decrease_listening_volume 8=admin_toggle_conference_lock 9=decrease_talking_volume *=admin_kick_last \#=participant_count === However, my user isn't hearing anything. MoH does work otherwise. What am I missing? Thanks in advance, Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for better fax handling
of course you can query asterisk asterisk and look, if your fax is still running...: asterisk -rx "fax show sessions" lists you all acive fax sessions... yves Am 22.05.2018 um 12:19 schrieb D'Arcy Cain: On 2018-05-22 02:17 AM, Yves wrote: you could - use "global variables" - use the asterisk built in database Both of those seem difficult as the process is split between Asterisk and an external script. - mv the file to temporary folder _before_ faxing (would be the most easy solution as you already know how to mv a file via asterisk...) True. This or John Kiniston's idea of lock files could work. I guess I would need to have some process to move it back if it is still there after an hour or so in case something went wrong. The same sort of thing would be needed for John's solution as well. It sure would be nice if I could query Asterisk to see if the fax process was still running. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for better fax handling
you could - use "global variables" - use the asterisk built in database - mv the file to temporary folder _before_ faxing (would be the most easy solution as you already know how to mv a file via asterisk...) regards, yves Am 21.05.2018 um 19:49 schrieb D'Arcy Cain: I am having troubles with sending faxes. I hope someone can help me work out a better method. Basically we have a special address that our users can send to. It winds up on our Asterisk server which runs a Python script that parses the message for attachments and the phone number from the recipient address. The attachments are converted to TIFF and stored in a folder with various information encoded into the file name such as phone number and retry information. That all works fine. A separate process picks up the files and sends them using an AMI script like this: Action: Originate Channel: SIP/provider/%(destination)s Context: LocalSets CallerID: Vybe Consulting Inc Fax Service <6475551212> Exten: sendfax Priority: 1 Timeout: 3 Variable: faxfile=%(faxfile)s Variable: uid=%(uid)s Variable: destination=%(destination)s Variable: sender_name=Vybe Consulting Inc Fax Service Variable: sender_num=6475551212 It then renames the file encoding the next retry time and incrementing the number of retries. The same script checks for files in a success folder and sends the users a confirmation message that the fax was sent. The files are moved into the success folder by Asterisk using this dialplan: sendfax,1,Verbose(0,FAX ${faxfile} to ${destination}) same => n,Set(FAXOPT(headerinfo)=${sender_name}) same => n,Set(FAXOPT(localstationid)=${sender_num}) same => n,SendFax(${faxfile},d) same => n,Set(STATUS=Status: ${FAXOPT(status)}) same => n,Set(STATUS=${STATUS}\nRemote ID: ${FAXOPT(remotestationid)}) same => n,Set(STATUS=${STATUS}\nMaxrate: ${FAXOPT(maxrate)}) same => n,Set(STATUS=${STATUS}\nMinrate: ${FAXOPT(minrate)}) same => n,Set(STATUS=${STATUS}\nECM: ${FAXOPT(ecm)}) same => n,Set(STATUS=${STATUS}\nnumber of pages: ${FAXOPT(pages)}) same => n,Set(STATUS=${STATUS}\nRate: ${FAXOPT(rate)}) same => n,Set(STATUS=${STATUS}\nResolution: ${FAXOPT(resolution)}) same => n,GotoIf($["${FAXOPT(status)}" = "SUCCESS"]?faxok) same => n,Set(STATUS=${STATUS}\nError: ${FAXOPT(error)}) same => n(faxok),Verbose(0,FAX ${destination} Status (S): ${STATUS}) same => n,Set(FAXNAME=${CUT(faxfile,/,6)}) same => n,Set(FILE(/fax_status/${FAXNAME})=${STATUS}) same => n,GotoIf($["${FAXOPT(status)}" != "SUCCESS"]?faxfail) same => n,System(/bin/mv '${faxfile}' '/fax_success/${FAXNAME}') same => n,Set(CDR(userfield)=${destination}) same => n,Verbose(0,FAX to ${destination} charged to ${uid}) same => n(faxfail),Verbose(0,FAX ${destination} Status (F): ${STATUS}) same => n,Hangup() My problem is that if the faxes get too big it starts sending it again before the previous one has finished. I can't raise the retry limit too far because sometimes the receiver is busy and we have to retry in a reasonable time. Is there a way to get a token from the AMI script that I can use to determine later if Asterisk is still busy with the fax before I try sending it again? Or, am I approaching this all wrong? Is there a better method of doing this? I am running Asterisk 13.19.0 on NetBSD/amd64 7.1.0. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pcapsipdump or general sip debug question - the solution
Hi, i know about this feature and use it a lot... my question was, how to get pcapsipdebug to generate only one file... BUT... meanwhile I found out how to accomplish this easy task. 1.) open first pcap file in wireshark 2.) open second pcap file in wireshark using the menu "file -> merge" 3.) go to "telephony -> sip flows" 4.) select the two "legs" of the call 5.) klick button "flow sequence" et voilà... one ladder diagram exactly the way I needed it thanks anyways, yves Am 17.01.2017 um 12:34 schrieb Jean Aunis: Hello, There is a built-in tool in Wireshark for this : menu Telephony => Voip Calls, the select your call and click on "Flow Sequence". Best regards Jean Aunis Le 17/01/2017 à 12:27, Yves a écrit : Hi, I am using pcapsipdump for debugging sip calls. when I have to debug a call, pcapsipdump generates two files per call... one for the sip dialog between the client (softphone) and the server (asterisk) and one for the sip dialog between the server (asterisk) and the sip registrar... is there a way to get this into one file ? the objective is to see both sides of the call in a single ladder diagram or just to have more comfort in analyzing the full flow within wireshark. If this is not possible, is there a free tool for sip (together with rtp) debugging that is able to catch the full sip flow between both ends of one call in a single file (per call) with pcap compatibility (including the rtp packets)? thank you yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pcapsipdump or general sip debug question
Hi, I am using pcapsipdump for debugging sip calls. when I have to debug a call, pcapsipdump generates two files per call... one for the sip dialog between the client (softphone) and the server (asterisk) and one for the sip dialog between the server (asterisk) and the sip registrar... is there a way to get this into one file ? the objective is to see both sides of the call in a single ladder diagram or just to have more comfort in analyzing the full flow within wireshark. If this is not possible, is there a free tool for sip (together with rtp) debugging that is able to catch the full sip flow between both ends of one call in a single file (per call) with pcap compatibility (including the rtp packets)? thank you yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SoundStation IP 6000 does not register
sorry... typo the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am 21.12.2016 um 15:13 schrieb Mauricio Tavares: On Wed, Dec 21, 2016 at 7:50 AM, Yves <yves...@gmx.de> wrote: Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves I am a bit confused: is your problematic phone's IP 192.168.0.13 (what the error log is reporting below) or 192.168.1.13? Am 21.12.2016 um 13:34 schrieb Mark Wiater: Yves, Didn't you say that AsteriskServer: 192.168.1.211 SIP-user: 165 ? On 12/21/2016 4:24 AM, Yves wrote: . It is sure for 100% that there is no firewall or something else mangeling in between... another Hardphone works as expected using the same Netzworkcable on the same Networkplug with UDP on Port 5060... This other hardphone, what IP does it have? 50.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask 255.255.255.0 The line above suggests to me that your phone and your asterisk server are on a different network, there has to be something that routes between those two networks. Often what routes, can firewall. 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 Temporarily not available Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SoundStation IP 6000 does not register
Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34 schrieb Mark Wiater: Yves, Didn't you say that AsteriskServer: 192.168.1.211 SIP-user: 165 ? On 12/21/2016 4:24 AM, Yves wrote: . It is sure for 100% that there is no firewall or something else mangeling in between... another Hardphone works as expected using the same Netzworkcable on the same Networkplug with UDP on Port 5060... This other hardphone, what IP does it have? 50.848|cfg |*|03|RT|Primary IP changed to 192.168.0.13 subnet mask 255.255.255.0 The line above suggests to me that your phone and your asterisk server are on a different network, there has to be something that routes between those two networks. Often what routes, can firewall. 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 Temporarily not available Mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom SoundStation IP 6000 does not register
Hi, I do not have a switch to mirror the traffic... I am only remotely connected to the office, where all is set up. I have full control over asterisk and the phone and I tcpdumped the traffic coming from the phone. The weird thing is... if I configure the SIP-Server Setting to use TCP on Port 80, I see REGISTER requests. If I configure to use UDP only on Port 5060, I do not see nothing at all... not a single Request coming from the phone... and, yes... It is sure for 100% that there is no firewall or something else mangeling in between... another Hardphone works as expected using the same Netzworkcable on the same Networkplug with UDP on Port 5060... Meanwhile I tried all available firmware-Versions, with and without provisioning. I am wondering about downloads, the phone is trying to receive from downloads.polycom.com that constantly fail (yes, these files do not exists there, the phone can communicate with the internet...) On the other hand, I don´t think that this has something to do with the problem, as the phone tries to REGISTER when I use TCP / 80 Olivier, would you mind and mail me your config-files and some screenshots from the phone-webconfig? Which software-versions are you using? thank you, yves if someone wants to take a look at the phone-logs: boot-log 02.335|so |*|01|-- Initial log entry -- 02.335|so |*|01|+++ Note that Updater log times are in GMT +++ 02.335|boot |*|01|Initial log entry. Current logging level 3 02.335|copy |*|01|Initial log entry. Current logging level 3 02.335|utilm|*|01|Initial log entry. Current logging level 4 02.335|hw |*|01|Initial log entry. Current logging level 4 02.335|ethf |*|01|Initial log entry. Current logging level 4 02.335|dns |*|01|Initial log entry. Current logging level 3 02.335|curl |*|01|Initial log entry. Current logging level 3 02.335|sec |*|01|Initial log entry. Current logging level 4 02.641|wdog |*|01|Initial log entry. Current logging level 4 02.641|lldp |*|01|Initial log entry. Current logging level 3 02.641|cdp |*|01|Initial log entry. Current logging level 3 02.641|key |*|01|Initial log entry. Current logging level 4 02.642|so |3|01|Platform: Model=SoundStation IP 6000, Assembly=3111-15600-001 Rev=W Region= 02.642|so |3|01|Platform: Board=3111-15600-001 B 0 02.642|so |3|01|Platform: MAC=0004f2070cd3 02.643|so |3|01|Platform: BootBlock=3.0.4.0001 (15600-001) 11-Jul-12 08:53 02.644|so |*|01|Platform: BootL1=Standalone.0008 26-Feb-08 14:11:56 02.644|so |3|01|Application, main: Label=Updater, Version=Azurite 5.0.5.2324 09-Dec-13 15:31 02.644|so |3|01|Application, main: P/N=-Y-YYY 02.644|log |*|01|Install file upload callback for 'Updater' 02.644|app1 |*|01|Initial log entry. Current logging level 3 02.645|cfg |*|01|Initial log entry. Current logging level 2 02.651|app1 |3|01|Application, load: Type=SIP, Version=4.0.4.2906 18-Apr-13 01:11 02.652|boot |*|01|Using TFFS for flash load 02.652|boot |*|01|Code length: 0x0097A585 02.652|boot |*|01|Code checksum: 0x4B86ABFB 03.631|so |3|01|Link status is Net up Speed 100 full Duplex. 17.497|app1 |4|01|Loaded application sip.ld from local system successfully. App-log 001139.870|app1 |*|03|Manual Reboot 001139.870|so |5|03|soAudioChannel compiledOffsetsApply error: unrecognized verAudio 11 for headset 001140.026|so |*|03|SoNcasC::procMsg: Client service shutdown complete 001144.025|wdog |*|03|Watchdog Expired: tSup 04.975|log |*|03|-- Initial log entry -- 04.975|so |*|03|Platform: Model=SoundStation IP 6000, Assembly=3111-15600-001 Rev=W Region= 04.975|so |*|03|Platform: Interfaceeth0 MAC=0004f2070cd3 04.977|so |*|03|Platform: BootBlock=3.0.4.0001 (15600-001) 11-Jul-12 08:53 04.977|so |*|03|Platform: BootL1=Standalone.0008 26-Feb-08 14:11:56 04.977|so |*|03|Platform: Updater=5.0.5.2324 09-Dec-13 15:31 04.977|so |*|03|Application, main: Label=SIP, Version=Mink 4.0.4.2906 18-Apr-13 01:11 04.977|so |*|03|Application, main: P/N=3150-11530-404 04.977|rdisk|*|03|RAM disk created, size: 8,388,608 bytes 04.978|ocsp |*|03|O.C.S.P. Enabled = 0 04.978|tls |*|03|Initial log entry. Current logging level 4 04.998|pmt |*|03|Initial log entry. Current logging level 4 04.998|wdog |*|03|Initial log entry. Current logging level 4 04.998|ethf |*|03|Initial log entry. Current logging level 4 04.998|hw |*|03|Initial log entry. Current logging level 4 04.998|ares |*|03|Initial log entry. Current logging level 4 04.998|dns |*|03|Initial log entry. Current logging level 4 04.998|cfg |*|03|Initial log entry. Current logging level 4 04.998|dot1x|*|03|Initial log entry. Current logging level 4 05.000|cfg |*|03|RT|Network eth0 link went up 05.000|cfg |*|03|RT
[asterisk-users] Polycom SoundStation IP 6000 does not register
Hi, I am pulling my hair for days now... I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register with my Asterisk. There are no SIP Packets arriving at my asterisk at all... and it has nothing to do with a firewall or similar... Simple Question: Does anybody have a running SoundStation IP 6000 registerd with asterisk? If so... would you please be so kind to tell me whats wrong with my setup? AsteriskServer: 192.168.1.211 SIP-user: 165 (the SIP-Settings on asterisk-side are OK, tested with a normal Softphone... registering and placing calls is no problem...) The phone-log only says: "Registration failed User: 165, Error Code:480 Temporarily not available" I tried with newest firmware, resetting to factory 100 times, using a provisionig file (which the SoundStation correctly downloads) but it is always the same... the SoundStation does not contact the asterisk for registering... Phoneversion: Telefoninformationen Telefonmodell SoundStation IP 6000 Teilenummer 3111-15600-001 Rev:W MAC-Adresse 00:04:F2:07:0C:D3 IP-Adresse 192.168.0.13 UC-Softwareversion 4.0.11.0583 BootROM-Softwareversion 5.0.5.2324 I can ping the phone from the asterisk, the phone can reach the asterisk server (as it downloads the tftp files, if used with a provisioning profile), so the route and everything is correct... I even connected another Hardphone on the same cable that stuck in the Polycom... no problem... the other phone can register and works, so there is really no cable or firewall related problem here... it must be a setting! thank you -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?
ok, thank you... then I´ll take it as it is cheers, yves Am 18.12.2016 um 13:15 schrieb Larry Moore: Hi, I haven't found anything definitive however I expect the TSI that is sent during initial fax call establishment is stored by the receiving terminal, see pages 28 & 29 of the English version of the document at https://www.itu.int/rec/T-REC-T.30-200509-I/en , I expect the header, which will include the TSI, is all part of the image (Tagline in HylaFAX) and not stored separately on the receiving terminal. Cheers, Larry. On 18/12/2016 6:20 PM, Yves wrote: Hi, thanks for your answer. Unfortunately this is, what I already know. I was wondering, why it is possible to set ID and Header for an outgoing fax (which will then in turn be inserted via asterisk on top of the transferred "image") , while it seems to not be possible to get the Header from a received fax (only the id), although it is present in the faxdocument. The ID is also present in the faxdocument and there does a Faxopt(remotestationid) exist... so I thought, this info must be transferred not only binary within the "image", but also within the "meta-data" / protocol-data of the fax (within the TSI) otherwise asterisk must do some kind of ocr to get the ID, what it definitely does not... btw... when using sendfax, asterisk inserts the date, the id, the header and the pagenum on top of each faxpage... someone knows how to modify some settings like font, position, and so on? thanks, yves Am 18.12.2016 um 00:02 schrieb Larry Moore: The list of options available are listed here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT It doesn't appear that a received header is available unless it is written into the 'headerinfo' variable after it is received, I haven't checked for this. From my days working with fax machines, the header could be inserted in the line the TSI is on or in the image being transmitted, if you receive a fax that has been sent to you with the latter set, then the 'headerinfo' will not be of any use. Perhaps someone with more knowledge may be able to explain this better. A quick Google search for 'fax header outside of tsi' will provide a list of manuals, here's one - http://manuals.konicaminolta.eu/bizhub-C554-C454-C364-C284-C224/EN/contents/sh3_378.html#qitem13 Expand the line for To specify the position of Header Position printed on a sent fax ([Header Position]) Larry. On 18/12/2016 4:30 AM, Yves wrote: Hi, I am using asterisk 11.8 in combination with spandsp to send and receive T38 Faxes. All works fine, but I do not know how to get the remoteheader from the fax I receive. When I send a fax, there are Faxopts to set the localstationid and the headerinfo, but for receiving, there seems to only exist the Faxopts remotestationid but for sure on any fax I receive there is a remoteheaderinfo besides the remotestationid... it is on the tiff-file, but I need this info in a channel-variable... Does anybody know how to get the remoteheaderinfo for a received fax? thanks yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Fax Receive - how to get the remoteheader?
Hi, thanks for your answer. Unfortunately this is, what I already know. I was wondering, why it is possible to set ID and Header for an outgoing fax (which will then in turn be inserted via asterisk on top of the transferred "image") , while it seems to not be possible to get the Header from a received fax (only the id), although it is present in the faxdocument. The ID is also present in the faxdocument and there does a Faxopt(remotestationid) exist... so I thought, this info must be transferred not only binary within the "image", but also within the "meta-data" / protocol-data of the fax (within the TSI) otherwise asterisk must do some kind of ocr to get the ID, what it definitely does not... btw... when using sendfax, asterisk inserts the date, the id, the header and the pagenum on top of each faxpage... someone knows how to modify some settings like font, position, and so on? thanks, yves Am 18.12.2016 um 00:02 schrieb Larry Moore: The list of options available are listed here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_FAXOPT It doesn't appear that a received header is available unless it is written into the 'headerinfo' variable after it is received, I haven't checked for this. From my days working with fax machines, the header could be inserted in the line the TSI is on or in the image being transmitted, if you receive a fax that has been sent to you with the latter set, then the 'headerinfo' will not be of any use. Perhaps someone with more knowledge may be able to explain this better. A quick Google search for 'fax header outside of tsi' will provide a list of manuals, here's one - http://manuals.konicaminolta.eu/bizhub-C554-C454-C364-C284-C224/EN/contents/sh3_378.html#qitem13 Expand the line for To specify the position of Header Position printed on a sent fax ([Header Position]) Larry. On 18/12/2016 4:30 AM, Yves wrote: Hi, I am using asterisk 11.8 in combination with spandsp to send and receive T38 Faxes. All works fine, but I do not know how to get the remoteheader from the fax I receive. When I send a fax, there are Faxopts to set the localstationid and the headerinfo, but for receiving, there seems to only exist the Faxopts remotestationid but for sure on any fax I receive there is a remoteheaderinfo besides the remotestationid... it is on the tiff-file, but I need this info in a channel-variable... Does anybody know how to get the remoteheaderinfo for a received fax? thanks yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Fax Receive - how to get the remoteheader?
Hi, I am using asterisk 11.8 in combination with spandsp to send and receive T38 Faxes. All works fine, but I do not know how to get the remoteheader from the fax I receive. When I send a fax, there are Faxopts to set the localstationid and the headerinfo, but for receiving, there seems to only exist the Faxopts remotestationid but for sure on any fax I receive there is a remoteheaderinfo besides the remotestationid... it is on the tiff-file, but I need this info in a channel-variable... Does anybody know how to get the remoteheaderinfo for a received fax? thanks yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WhatsApp feature on Asterisk
Can anyone put light on whatsapp features and how it can be operated . What are the technology that need to be installed , Regards Yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial
Asterisk 1.8.23.0-1_centos5.go DAHDI Version: 2.6.1 Echo Canceller: HWEC On Wed, Jul 20, 2016 at 5:32 PM, A J Stiles <asterisk_l...@earthshod.co.uk> wrote: > On Wednesday 20 Jul 2016, Yves biganiro wrote: > > Hi all > > > > Hi,I'm facing a strange issue where by SANGOMA not detected by > goautodial > > system , > > Is this some kind of one-stop, pre-prepared distribution with Linux, > Asterisk, > DAHDI, a web server and some custom scripts, that all installs from one > place? > > We really need to know your Asterisk and DAHDI versions. > > Type in a root terminal, > > # asterisk -V > > and note the version number displayed (it will be on the first line). > Then > enter > > *CLI> dahdi show version > > and note the DAHDI version displayed. > > -- > AJS > > Note: Originating address only accepts e-mail from list! If replying off- > list, change address to asterisk1list at earthshod dot co dot uk . > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Yves Biganiro Senior IT Consultant - independent Tel +250727612605 ##A tech entrepreneur and web developer, Passionate about technology with working experience in web development.## -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial
I have forcefully installed everything but it says that the card is not found. On Wed, Jul 20, 2016 at 5:05 PM, Yves biganiro <yves.bigan...@gmail.com> wrote: > Hi all > > Hi,I'm facing a strange issue where by SANGOMA not detected by > goautodial system , Thats the problem : > Configuring ISDN BRI cards [A500/B700] > > > No Sangoma ISDN BRI cards detected > > Press any key to continue: > > Configuring GSM cards [W400] > > > No Sangoma GSM cards detected > > > > regards > > > > > -- Yves Biganiro Senior IT Consultant - independent Tel +250727612605 ##A tech entrepreneur and web developer, Passionate about technology with working experience in web development.## -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No Sangoma ISDN BRI cards detected by goautodial
Hi all Hi,I'm facing a strange issue where by SANGOMA not detected by goautodial system , Thats the problem : Configuring ISDN BRI cards [A500/B700] No Sangoma ISDN BRI cards detected Press any key to continue: Configuring GSM cards [W400] No Sangoma GSM cards detected regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] open source pbx free
Anyone have any experience running an open source pbx and call center solution?Need to start a call center of 10 users and i need help I have already installer a server with Ubuntu Server 14.04 , E1 installed Please advice me how to process from here Regards Yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] my dahdi dont'n start
Hello, I was faced with this problem, it is enough to place subdirectory under ./tools installation dahdi when compiling and run make install-config it should work. we must have : mkdir -p / etc / dahdi mkdir -p /etc/modprobe.d install -m644 xpp / genconf_parameters / etc / dahdi / genconf_parameters install -m644 init.conf.sample /etc/dahdi/init.conf install -m644 blacklist.sample /etc/modprobe.d/dahdi-blacklist.conf install -m644 modprobe.conf.sample /etc/modprobe.d/dahdi.conf make -f ./Makefile.legacy top_srcdir =. srcdir =. config make [1]: Entering directory '/usr/src/dahdi-linux-complete-2.11.1+2.11.1/tools' install -D dahdi.init /etc/init.d/dahdi /usr/sbin/update-rc.d dahdi defaults 15 30 DAHDI has-been configured. Le 28/04/2016 16:37, A J Stiles a écrit : On Thursday 28 Apr 2016, Mamadou NGOM wrote: Hello, it doesn't work my dahdi yet .for information, i use debian 8 . I put the file dahdi.bash in /etc/init.d and I gave it the permission 755 but i have the same error: bash: /etc/init.d/dahdi: No such file or directory You need to name the file just "dahdi", not "dahdi.bash"; because the command "service dahdi start" is looking for a file just called "dahdi". If you run # mv /etc/init.d/dahdi.bash /etc/init.d/dahdi then # service dahdi start should work. You probably also need to run # update-rc.d dahdi defaults to ensure it starts up everytime the computer is booted up. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?
Le 06/04/2016 18:12, Markos Vakondios a écrit : Good evening, My English is limited but if I can help. We install Asterisk Version 13.1 on VmWare with Debian 8.2, no problem since June 2015, currently I have tested on Unbutu 14.04 but problem with network-manager (problem of stability with Asterisk 1.8.32 and difficulty with routing network-manager). I also installed Asterisk on KVM (Debian 8.2) no problem (but not test with dahdi) without particular problem. here is my little opinion Hello everyone Proxmox and KVM on Ubuntu On Wednesday, 6 April 2016, Ryan, Travis> wrote: What is the best virtual server tech (and most stable, etc) to use for a asterisk virtual hosting environment? I have a client that wants to do virtual hosting of Asterisk (only SIP or IAX, no PRI, etc) and I’m wondering if Xen or something else would be best? We’d like to stay away from the costs of VMWare if possible. Thanks! Travis // -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The To header was truncated in call... Whats this means?
I have seen these messages only on asterisk boxes that are open to public and I think this may have something to do with sip-attacks... I´d recommend some wiresharking or at least sip debugging... yves Am 07.01.2016 um 21:23 schrieb Vitor Mazuco: Hi everybody, My Asterisk, all time appear this log [Jan 7 15:37:04] ERROR[1174] chan_sip.c: The To header was truncated in call '6c66e5b6058ae257003c0f7e778da0fe@191.x'. This call setup will fail. [Jan 7 15:37:18] ERROR[1174] chan_sip.c: The To header was truncated in call '18e0a12e434364254b0cc2e52d20755b@191.x. This call setup will fail. ... Whats this massege means? Thanks. --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. https://www.avast.com/antivirus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] placing calls with linphone.org SIP account
Happy new year! maybe off-topic... but maybe someone _knows_ a solution. I have a free SIP Account at linphone.org... calling other linphone.org users via SIP and receiving SIP calls from other users registered at linphone.org is no problem... just "dial" the username... BUT... as far as I understand the documentation, linphone.org offers internet wide SIP Calls... so... how can I call other users registered at other SIP-Providers? I tried all well-known SIP URI Syntaxes but none worked... does anyone reliably know, if it is possible at all and if so, what is the dialstring looking like? (I am trying with zoiper softphone) Unfortunately there is no support-email-address for linphone.org users... thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. https://www.avast.com/antivirus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No QueueCallerJoin Event...
Hi, I am using asterisk 13.5.0 and although my AMI-user has read=all and write=all permissions, I don´t get any QueueCallerJoin Events fired, when a new caller calls into a Queue... Strange enough, a QueueCallerAbandoned Event is fired, when the caller hangs up without beeing connected to an agent... Is it a bug or am I missing something? regards, Yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. https://www.avast.com/antivirus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-java is dead?
No, its not dead and mails to the asterisk-java-list become replied. regards, yves Am 18.09.2015 um 02:35 schrieb symack: Hello Everyone, I am trying to make use of asterisk-java live and had some questions for the mailing list however, it does not seem like it's an active mailing list? Is the project dead? Thanks, Nick --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. https://www.avast.com/antivirus -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call failed... but why? What means SIP_ALREADYGONE?
Hi, I have watched a phenomen, that I can not explain... maybe one of you can see the reason why the call failed, and if the cause is the Snom Hardphone, or the asterisk, or the SIP-Provider... the debug log given below is all I have... What does Setting SIP_ALREADYGONE on dialog.. mean? thanks for watching, yves SIP Phone 110 (callerid 061444018110) tried to call the external Phone Number 0616677823 and gets an hangup after 2 seconds. Another try immediately after the failed call goes fine. The failed call did not arrive at the destination. [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Begin: parsing SIP Supported: timer, 100rel, replaces, from-change [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -timer- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: timer [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -100rel- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: 100rel [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -replaces- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: replaces [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Found SIP option: -from-change- [Feb 12 10:00:11] DEBUG[1567][C-380e] sip/reqresp_parser.c: Matched SIP option: from-change [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.0.165:3072 [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Connection okay. [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '00616677823' AND h ost = 'dynamic' [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Connection okay. [Feb 12 10:00:11] DEBUG[1567][C-380e] res_config_mysql.c: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_peers WHERE name = '00616677823' [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Stopping retransmission on '9a6bdc548d19-goay25ioz0nd' of Response 1: Match Found [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7f2a74158788' [Feb 12 10:00:11] DEBUG[1567][C-380e] res_rtp_asterisk.c: Allocated port 19528 for RTP instance '0x7f2a74158788' [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: RTP instance '0x7f2a74158788' is setup and ready to go [Feb 12 10:00:11] DEBUG[1567][C-380e] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7f2a74158788' [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Setting NAT on RTP to On [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP o=root 871055034 871055034 IN IP4 192.168.0.165... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP s=call... UNSUPPORTED OR FAILED. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.0.165... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 9 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 0 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 8 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 99 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 108 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 18 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] rtp_engine.c: Setting payload 101 based on m type on 0x7f2a80b1a620 [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:99 G726-32/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:108 AAL2-G726-32/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK. [Feb 12 10:00:11] DEBUG[1567][C-380e
Re: [asterisk-users] SIP call drops after 32 seconds, but only when....
Hi, I know this Bug,,, at least when you´re talking about x-lite 3... quite annoying, but if you know it... so no... its not the phone... tested with zoiper and 3cx ... both work...but the problem occurs ONLY, as soon as I register at more than one registrar... yves Am 22.11.2014 um 19:19 schrieb Ron Wheeler: You might check your phones as well. We had this problem early on with a softphone and it was a setting in the phone that was set to hang up after 30 seconds of inactivity in case of network disruption. For some reason it was detecting network disruption in every call even when the calls were proceeding normally. Unchecking this box solved the problem. It may not be related to your problem but if it is the cause, you will spend a lot of time trying to fix this in Asterisk. :-D At least I did! On the bright side, it does force people to get point in a hurry! Ron On 22/11/2014 12:50 PM, Eric Wieling wrote: Try setting directmedia=no in sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? thanks for taking part... I don´t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP call drops after 32 seconds, but only when....
Hi, the useragents nothing to do with the problem... i tried numeric, alpha and alphanumeric... no difference. they work all as long as I only use ONE registrar... as soon as I register at more than one registrar... the line drops after 32 seconds really strange. yves Am 22.11.2014 um 19:01 schrieb Rafael Visser: Hi Yves.. This may be silly... but what is the useragent of your sip configuration? In the case that useragent has some special characters like (., please remove it and tell us if there is any change!!. Regards. rv 2014-11-22 14:50 GMT-03:00 Eric Wieling ewiel...@nyigc.com mailto:ewiel...@nyigc.com: Try setting directmedia=no in sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP call drops after 32 seconds, but only when Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? thanks for taking part... I don´t know... one is siptrunk.ovh.net http://siptrunk.ovh.net and the other one is sip.ovh.fr http://sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to set timerb in sip.conf
Hi list, I have tried to set the value for timerb in sip.conf, general section and in user-context... tried on asterisk 1.4 up to version 13... no success. The value for timerb remains unchanged. (reload, restart, reboot all does not help...) sip show settings always show 32000ms for timerB. How can I configure the timerb value? thx, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphone signals busy although it isn´t
Hi, I have written an click2dial application that rings an agent soft phone and connects the agent with a customer. very often I can see, that the agent softphones signal a busy back to server, although the phone is definitely hung up and the previous calls where handled normally. I testet 3cx Version 6 and X-Lite V1 up to V3... all show the same misbehaviour. I did a SIP Trace and can see, the phone replies with a busy on the invite... but I don´t know why. Has anybody experienced similar things and knows a reason or a workaround for this? I am working on a asterisk 11.7 using sip-realtime peers with mysql. thanks for reading, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP call drops after 32 seconds, but only when....
hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in between... I tried to work around this by increasing the settings for timerb... but I realized that asterisk does not care at all, what I set this value to... sip show settings always gives me 32000ms, and it does not make any difference if I configure timerb in the general context or in the phone context... any ideas? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP call drops after 32 seconds, but only when....
Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. Are both your servers behind the same NAT router? thanks for taking part... I don´t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren geprüft. http://www.avast.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best strategy to find and solve voice quality problems
Hi, in my company we use an asterisk installation with around 50 soft- and hardphones of all kind. From time 2 time the users (almost only Softphone users) report some voice qualities... mostly echoes. These problems do not occur on all PCs at the same time and since setup of our PBX almost any PC user has gotten these issues. When I come there to check, everything is fine again... and I can´s see anymore, what could have caused the problem... may it be a high network load, or a high cpu usage or whatever... I activated call recording to hear the quality after such missing-quality reports but every call I listened to showed no issues in the recording so I assume the problem is on the client side. Because it is not always the same user or the same PC I think it cannot be a misbehaviour like wrong headset usage or a problem of a single PC. What is the best strategy to find and solve these kind of problems? Are there any (free would be cool) tools that can monitor the pc-state (concerning at least network and cpu- / process usage) over a long period and display the results in an appropriate way? Is there a way under Windows XP / 7 to ensure Bandwidth for VoIP like QoS (google only showed me such settings for Lync or Windows Server machines...? Is there a way under Windows XP / 7 to ensure CPU-Bandwidth for Applications (like VoIP Clients)? Thanks for any hint, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to send dtmf after pause ?
This would be possible with an agi... the agi can wait for silence or 10 seconds, as u like and then play the dtmf tones and bridge the call to your extension afterwards. yves Am 07.06.2013 17:51, schrieb Sean Darcy: I'm trying to call a conference service, wait 10 seconds, then send the passcode. I've tried ww: Dial(SIP/18005551212ww12345#@sip.com,60,r) The sip channel didn't like that. Added 'p' , still no help. I tried D: Dial(SIP/18005551...@sip.com,60,rD(12345#) The dtmf is sent too soon. I tried inserting 'ww' but that was just sent. I tried G: exten = 234.1.Dial(SIP/18005551...@sip.com,60,rG(next)) same=n(next),Wait(10) same=n,SendDTMF(12345#) but that didn't work at all, This is a common use case. There must be some simple answer I'm missing. Thanks for any help. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?
looks yummy indeed... but how does it interact with an asterisk? phono uses afaiu voxeo-cloud to make place calls, send sms and so on... I do not see a way to use phono without their cloud services, not did I see any hint about charges for calls... yves Am 03.06.2013 12:34, schrieb Lenz Emilitri: Looks yummy! http://phono.com/webrtc 2013/5/31 Adnan 112linuxstockh...@gmail.com mailto:112linuxstockh...@gmail.com Voxeo/Phono webrtc. /Adnan On Fri, May 31, 2013 at 1:53 PM, Lenz Emilitri lenz.lo...@gmail.com mailto:lenz.lo...@gmail.com wrote: Hi All, I wonder if any of you has some suggestions on which WebRTC client/softphone to use for a click-to-dial, webpage hosted solution. Any suggestions? Thanks l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me understand these log messages
... an anonyous (not registerted) sip user from 188.161.238.232 was trying to initiate a call to 9725955 and so on... you could enable sip tracing to get more information. maybe you should change the 'allowguest' option in sip.conf..? regards, yves Am 31.05.2013 23:57, schrieb Chris Gentle: OK, I need a bit of help here. I'm configuring a new Asterisk 11 system and I accidentally let my firewall rules drop for a day or so. When I logged in today, I found messages like the ones below on my asterisk console. Obviously somebody was trying to take advantage of my carelessness. So can someone explain what would cause these types of messages to show up on my console? I understand that my iptables would have stopped this but I'm just trying to understand more about the problem. What other settings might have stopped this? Fail2ban was running but there were no failed registration type messages that would have triggered it. [May 31 01:47:40] NOTICE[2544][C-0001] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '972595595767' rejected because extension not found in context 'default'. [May 31 01:47:40] VERBOSE[2544][C-0002] netsock2.c: == Using SIP RTP CoS mark 5 [May 31 01:47:40] NOTICE[2544][C-0002] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '00972595595767' rejected because extension not found in context 'default'. [May 31 01:47:41] VERBOSE[2544][C-0003] netsock2.c: == Using SIP RTP CoS mark 5 [May 31 01:47:41] NOTICE[2544][C-0003] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '000972595595767' rejected because extension not found in context 'default'. [May 31 01:47:41] VERBOSE[2544][C-0004] netsock2.c: == Using SIP RTP CoS mark 5 [May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from '' (188.161.238.232:28203) to extension '011972595595767' rejected because extension not found in context 'default'. snip -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Executing a dynamic sequence of applications
Hi, I would recommend an AGI-script or a realtime dialplan for this purpose. yves Am 30.05.2013 11:46, schrieb Grant Bagdasarian: Hello, I'm researching the possibilities of multiple communication platforms like Asterisk and FreeSwitch for handling a dynamic sequence of applications to execute, like Playback, Read, etc. This only applies to originating a call from an external application by using the AMI Manager and the Originate action. I need to know the following: 1)Does the Originate action support multiple Application keys? If so, how does it handle the order in which they're added to the Originate action? 2)If it does not support multiple Application keys, I'll have to instruct the Originate action to enter a context in the dialplan, and pass the sequence of applications in its Variable key. How would I configure the dialplan context to dynamically handle the sequence of applications to execute? I was thinking of creating a separate priority label for each required Application and have each application in the Variable key routed to the correct priority label. Is this possible? Are there alternatives for doing what I require? Regards, Grant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Wanpipe Driver
- solved - it turned out that libpri was not compiled correctly... and... Asgars comment about group systax is correct. thx regards, yves Am 13.05.2013 13:21, schrieb Yves A.: that was the syntax before 1.8 or 11.x I think... what about pseudo? yves Am 13.05.2013 13:16, schrieb Asghar Mohammad: Dial(DAHDI/i0/number, is it not Dial(DAHDI/r0/number or Dial(DAHDI/R0/number or Dial(DAHDI/g0/number or Dial(DAHDI/G0/number? On Mon, May 13, 2013 at 12:53 PM, Yves A. yves...@gmx.de mailto:yves...@gmx.de wrote: mmh... actually supportline is closed... why proceeds the call to dahdi/pseudo-?? i have never seen this before... thx., yves Am 13.05.2013 11:42, schrieb Duncan Turnbull: We have had challenges with the latest kernel versions on Ubuntu and sangoma wanpipe drivers An older kernel - no problem, latest ones, sometime risky. There are release notes on their site stating the supported versions so it might pay to check that But if it compiled ok it might be something else Sangoma support will dial in and help you if you ask them Cheers Duncan On 13/05/2013, at 9:29 PM, Yves A. yves...@gmx.de mailto:yves...@gmx.de wrote: Hi, I migrated from asterisk 1.6 to 11.3. The Server has a Sangoma A104 quadPri card installed. OS is a fresh installed Ubuntu 12.04 64bit libpri, dahdi etc. all latest releases.. Sangoma says... driver is compatible with ANY asterisk version... I tried driver 3.5.8... Setup ended with error. I tried (latest) driver 7.0.1 Setup went through, Asterisk is showing dahdi channels... all fine I thought... but..: when dialing Dial(DAHDI/i0/number) it accepts the call, but generates a DAHDI/Pseudo channel and the call goes not into the PSTN... What am I doing wrong? Has anybody successfully compiled sangoma driver 7.0.1 in combination with an asterisk 11.3? thanks for hints, regards, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] amiDebugger - might make your life easier if you program through the AMI
thank you! such efforts for the community are always highly appreciated! - I´ll give it a try. regards, yves Am 13.05.2013 21:44, schrieb Lenz Emilitri: Hi all, I have been playing with the AMI quite a bit lately - mostly debugging WombatDialer in production, but that's a different story - and I have been frustrated by the lack of a simple way to interact CLI-like with the AMI itself. So I have decided to write something myself to make my life easier, or at least a bit less miserable. The result is a little webapp that you can use as a sort of CLI-frontend to the AMI itself. It is not pretty, but pretty much effective. So I thought I could share it and make someone else's life a bit easier. You can find it on https://github.com/l3nz/amiDebugger - if you just want to test-drive it get the WAR file an put it into some webapp container, e.g. Tomcat. Hope you'll like it. l. -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sangoma Wanpipe Driver
Hi, I migrated from asterisk 1.6 to 11.3. The Server has a Sangoma A104 quadPri card installed. OS is a fresh installed Ubuntu 12.04 64bit libpri, dahdi etc. all latest releases.. Sangoma says... driver is compatible with ANY asterisk version... I tried driver 3.5.8... Setup ended with error. I tried (latest) driver 7.0.1 Setup went through, Asterisk is showing dahdi channels... all fine I thought... but..: when dialing Dial(DAHDI/i0/number) it accepts the call, but generates a DAHDI/Pseudo channel and the call goes not into the PSTN... What am I doing wrong? Has anybody successfully compiled sangoma driver 7.0.1 in combination with an asterisk 11.3? thanks for hints, regards, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Wanpipe Driver
mmh... actually supportline is closed... why proceeds the call to dahdi/pseudo-?? i have never seen this before... thx., yves Am 13.05.2013 11:42, schrieb Duncan Turnbull: We have had challenges with the latest kernel versions on Ubuntu and sangoma wanpipe drivers An older kernel - no problem, latest ones, sometime risky. There are release notes on their site stating the supported versions so it might pay to check that But if it compiled ok it might be something else Sangoma support will dial in and help you if you ask them Cheers Duncan On 13/05/2013, at 9:29 PM, Yves A. yves...@gmx.de wrote: Hi, I migrated from asterisk 1.6 to 11.3. The Server has a Sangoma A104 quadPri card installed. OS is a fresh installed Ubuntu 12.04 64bit libpri, dahdi etc. all latest releases.. Sangoma says... driver is compatible with ANY asterisk version... I tried driver 3.5.8... Setup ended with error. I tried (latest) driver 7.0.1 Setup went through, Asterisk is showing dahdi channels... all fine I thought... but..: when dialing Dial(DAHDI/i0/number) it accepts the call, but generates a DAHDI/Pseudo channel and the call goes not into the PSTN... What am I doing wrong? Has anybody successfully compiled sangoma driver 7.0.1 in combination with an asterisk 11.3? thanks for hints, regards, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Wanpipe Driver
that was the syntax before 1.8 or 11.x I think... what about pseudo? yves Am 13.05.2013 13:16, schrieb Asghar Mohammad: Dial(DAHDI/i0/number, is it not Dial(DAHDI/r0/number or Dial(DAHDI/R0/number or Dial(DAHDI/g0/number or Dial(DAHDI/G0/number? On Mon, May 13, 2013 at 12:53 PM, Yves A. yves...@gmx.de mailto:yves...@gmx.de wrote: mmh... actually supportline is closed... why proceeds the call to dahdi/pseudo-?? i have never seen this before... thx., yves Am 13.05.2013 11:42, schrieb Duncan Turnbull: We have had challenges with the latest kernel versions on Ubuntu and sangoma wanpipe drivers An older kernel - no problem, latest ones, sometime risky. There are release notes on their site stating the supported versions so it might pay to check that But if it compiled ok it might be something else Sangoma support will dial in and help you if you ask them Cheers Duncan On 13/05/2013, at 9:29 PM, Yves A. yves...@gmx.de mailto:yves...@gmx.de wrote: Hi, I migrated from asterisk 1.6 to 11.3. The Server has a Sangoma A104 quadPri card installed. OS is a fresh installed Ubuntu 12.04 64bit libpri, dahdi etc. all latest releases.. Sangoma says... driver is compatible with ANY asterisk version... I tried driver 3.5.8... Setup ended with error. I tried (latest) driver 7.0.1 Setup went through, Asterisk is showing dahdi channels... all fine I thought... but..: when dialing Dial(DAHDI/i0/number) it accepts the call, but generates a DAHDI/Pseudo channel and the call goes not into the PSTN... What am I doing wrong? Has anybody successfully compiled sangoma driver 7.0.1 in combination with an asterisk 11.3? thanks for hints, regards, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Building Asterisk 11.4.0-rc1 with PJSIP 2.1
hi, i would try to make a symlink... link the wrong folder to the correct one... yves Am 02.05.2013 23:34, schrieb James Mortensen: Hello, I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of 2.0 due to a crashing issue resulting from ICE. https://issues.asterisk.org/jira/browse/ASTERISK-21696 Currently, I'm systematically going through each Makefile in every directory in pjproject and changing the paths that exist in the pjproject 2.0 included with Asterisk, so that I can successfully build Asterisk. I'm using the Asterisk pjproject 2.1 port from here: https://github.com/asterisk/pjproject An example of the build errors I'm resolving one by one is this: make[2]: *** No rule to make target `../../pjlib/lib/libpj-x86_64-unknown-linux-gnu.a', needed by `../lib/libpjnath-x86_64-unknown-linux-gnu.a'. Stop. make[1]: *** [/mnt/src/asterisk-11.4.0-rc1/res/pjproject/pjnath/lib/libpjnath-x86_64-unknown-linux-gnu.a] Error 2 make: *** [res] Error 2 I'm editing the Makefiles and fixing the paths so Asterisk can find the target. For all the people out there smarter than me, is there a better way to go about this? I'm hoping upgrading PJSIP will resolve the crashing issue, and I'll continue going through Makefiles until someone smarter than me can enlighten me. Thank you for your help! -- James Mortensen Project Manager, VoiceCurve, Inc. 866-707-4590 james.morten...@voicecurve.com mailto:james.morten...@voicecurve.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for a way to do appointment reminders
Hi Brandon, as you are asking for professional help for a commercial project, I would recommend you to place a bounty. You can contact me directly if you want my professional help... I have developed exactly what you´re looking for and this solution is running in a high-call-volume installation without any issues. regards, yves Am 26.04.2013 03:55, schrieb Brandon Coale: Hello, My health care organization is looking for a way to do appointment reminders. We currently have staff members who spend part of each day manually calling patients to remind them of their upcoming appointments, and we would like to automate this process. Our electronic health record software would provide such information as the patient's name, phone number, and day and time of the appointment, and Asterisk could take this information and place an automated call to the patient. We would like the reminder call to use text-to-speech to personalize the call, such as We have an appointment reminder for [first name]. The appointment is on [date] at [time]. I am wondering if anyone has experience with using Asterisk for this type of application, and would be willing to share any details of how you implemented it? I am interested in any ideas, from very simple to feature-rich. We would be doing a new installation of Asterisk for this purpose, so we could use any version of Asterisk you would recommend. Thank you! Brandon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI DEBUG
hi, strange behaviour while trying to use pri debugging on asterisk 11.x ... please take a look: bas1104*CLI pri show version libpri version: 1.4.13 bas1104*CLI dahdi show version DAHDI Version: 2.6.1 Echo Canceller: HWEC bas1104*CLI help pri *pri intense debug span*no description available pri service disable channel Remove a channel from service pri service enable channel Return a channel to service *pri set debug {on|off*|hex|inte Enables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show channels Displays PRI channel information *pri show debug*Displays current PRI debug settings pri show spans Displays PRI span information pri show span Displays PRI span information pri show version Displays libpri version bas1104*CLI help dahdi dahdi destroy channel Destroy a channel dahdi restart Fully restart DAHDI channels dahdi set dnd Sets/resets DND (Do Not Disturb) mode on a channel dahdi set hwgain Set hardware gain on a channel dahdi set swgain Set software gain on a channel dahdi show cadences List cadences dahdi show channels [group|con Show active DAHDI channels dahdi show channel Show information on a channel dahdi show status Show all DAHDI cards status dahdi show version Show the DAHDI version in use / //currently all debug off:/ bas1104*CLI pri show debug Span 1: Debug: No Intense: No Span 2: Debug: No Intense: No Span 3: Debug: No Intense: No Span 4: Debug: No Intense: No / //switching it on (which currently works as expected)/ bas1104*CLI pri intense debug span 1 Enabled debugging on span 1 / // //oops, still shows no debug but it IS activated.../ bas1104*CLI pri show debug Span 1: Debug: No Intense: No Span 2: Debug: No Intense: No Span 3: Debug: No Intense: No Span 4: Debug: No Intense: No / //huh... how to disable it again? on some machines I can do so with pri no debug span nr but not here... gives same result (no// //such command) and debug is still enabled.../ bas1104*CLI pri set debug off No such command 'pri set debug off' (type 'core show help pri set' for other possible commands) bas1104*CLI so... whats the right way to disable pri debugging? thx, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.2.1 / dahdi destroy channel / asterisk crashes
Hi, I can reproduce your report (11.0.1, libpri 1.4.13, dahdi 2.6.1) and would say it is a bug... To remotely hang up a call use * **hangup request channel* where channel is the exact id of your channel as you would receive it via *core show channels* yves Am 11.04.2013 10:56, schrieb Thorsten Göllner: Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a Dial(DAHDI/g1/voicenumber,r). The call is bridged and everything is fine. dahdi show channels shows me, that channel 1 is used for the outcall. Then I try to hangup the outcall via dahdi destroy channel 1. Asterisk crahes immediatly. No message is logged (verbose is 10 and debug is 10). I get disconnected from the atserisk cli at this moment: vlr-3*CLI dahdi destroy channel 1 vlr-3*CLI Disconnected from Asterisk server Asterisk cleanly ending (0). Executing last minute cleanups voxi@vlr-3:/tmp$ Is this a bug or is this my fault? Best regards -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI DEBUG
thanks, that command syntax works. yves Am 11.04.2013 18:51, schrieb Richard Mudgett: - Original Message - hi, strange behaviour while trying to use pri debugging on asterisk 11.x ... please take a look: bas1104*CLI pri show version libpri version: 1.4.13 bas1104*CLI dahdi show version DAHDI Version: 2.6.1 Echo Canceller: HWEC bas1104*CLI help pri pri intense debug span no description available pri service disable channel Remove a channel from service pri service enable channel Return a channel to service pri set debug {on|off |hex|inte Enables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show channels Displays PRI channel information pri show debug Displays current PRI debug settings pri show spans Displays PRI span information pri show span Displays PRI span information pri show version Displays libpri version bas1104*CLI help dahdi dahdi destroy channel Destroy a channel dahdi restart Fully restart DAHDI channels dahdi set dnd Sets/resets DND (Do Not Disturb) mode on a channel dahdi set hwgain Set hardware gain on a channel dahdi set swgain Set software gain on a channel dahdi show cadences List cadences dahdi show channels [group|con Show active DAHDI channels dahdi show channel Show information on a channel dahdi show status Show all DAHDI cards status dahdi show version Show the DAHDI version in use currently all debug off: bas1104*CLI pri show debug Span 1: Debug: No Intense: No Span 2: Debug: No Intense: No Span 3: Debug: No Intense: No Span 4: Debug: No Intense: No switching it on (which currently works as expected) bas1104*CLI pri intense debug span 1 Enabled debugging on span 1 oops, still shows no debug but it IS activated... It activated a different mode of debug than what you expected because that command is an alias that was not updated. bas1104*CLI pri show debug Span 1: Debug: No Intense: No Span 2: Debug: No Intense: No Span 3: Debug: No Intense: No Span 4: Debug: No Intense: No huh... how to disable it again? on some machines I can do so with pri no debug span nr but not here... gives same result (no such command) and debug is still enabled... bas1104*CLI pri set debug off No such command 'pri set debug off' (type 'core show help pri set' for other possible commands) bas1104*CLI so... whats the right way to disable pri debugging? The correct command is pri set debug {on|off|intense} span x. The pri intense debug span x command is an alias for pri set debug 2 span x that didn't get updated when the real command was changed to pri set debug intense span x. This will show the help you need: bas1104*CLI help pri set debug off span Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi-channels.conf vs. chan_dahdi.conf
hi, chan_dahdi is some kind of generic config file, and dahdi-channels the config file where you configure your channels... so to say hardware specific. dahdi-channels.conf is normally a generated file which in turn is included by chan-dahdi... it makes sense to me to divide dahdi channel config in two files... but as one of them is included by the other you could merge them by hand... but remember you have then to edit it yourself if your hardware configuration changes (e.g. after adding a new card, as it was in your case...) so if your analog card requires drivers, install them or look in you /etc/dahdi/modules if you disabled the loading of the module for your newly added card. after this run dahdi_genconf and all should be set up atomagically... regards, yves Am 28.03.2013 14:44, schrieb Ken D'Ambrosio: Hey, all. Just added an analog card to our dual-T1 system... and clearly I'm doing something wrong. Less interested in having the specifics pointed out than in finding out how/why certain things work. So, really, three things: * What the bloody Hell is the difference between dahdi-channels.conf and chan_dahdi.conf? (And who thought it was a good idea to have two files with, apparently, different functionality, but very similar names?) * If I'm getting power to my analog phones, but no dial tone, which file should I be editing? * Likewise (and almost certainly related) if dahdi_cfg shows the channels, but dahdi show channels only shows my T1 spans, which file should I be editing? Could someone point me to some sample analog configs? Most of my searches have wound me up with GUI folks, and I'm just doing good ol-fashioned hand editing on an Ubuntu system. Thanks! -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING[28151] from CLI
Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... regards, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING[28151] from CLI
you have already listed the two config files for using zaptel. on first sight, they look ok to me (did not use zaptel for years now) maybe you should definitely comment out any span that is not in use... or do the opposite. i´ve seen this warning several times, but i cant remember it had anything to do with spans being configured but not used. it always had something to do with timing or even defective cards or cabling or even wrong settings on providers´ site. what changes were made to the system so that these warnings occur? or have they been visible from the very start? do they affect telefony (e.g. loss of calls, one side audio only etc.)? how much load (concurrent calls) is on the asterisk, does the warning occur periodically or only a few times? these are all questions you should ask yourself to help you find the answer yourself... it can be very frustrating sometimes, but for me, thats all i can tell about. regards, yves Am 27.03.2013 13:06, schrieb Salaheddine Elharit: thank you for your help ,but which configure script and when i can find this script ? in etc/asterisk best regards 2013/3/27 Thorsten Göllner t...@ovm-group.com mailto:t...@ovm-group.com You do use only span 1 and 6? So the other ports are not plugged? That is the cause for the warnings. I use a Sangoma E1-Card. The configure script gives me the option unused for any port. Maybe your configure script offers you the same option. Am 27.03.2013 11:54, schrieb Salaheddine Elharit: Hi i use 2 digium cards 1 card with 2 ports and the second card with 4 ports but actually i use just the span 1 and span 6 Asterisk 1.4-r110474M i use E1 ports zaptel.conf # Autogenerated by /usr/sbin/zapconf on Wed Feb 20 10:13:17 2013 -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS RED span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 # span=3,3,0,ccs,hdb3 # termtype: te # bchan=63-77,79-93 # dchan=78 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 # span=4,4,0,ccs,hdb3 # termtype: te # bchan=94-108,110-124 # dchan=109 # Span 5: TE2/1/1 T2XXP (PCI) Card 1 Span 1 span=5,5,0,ccs,hdb3 # termtype: te bchan=125-139,141-155 dchan=140 # Span 6: TE2/1/2 T2XXP (PCI) Card 1 Span 2 span=6,6,0,ccs,hdb3 # termtype: te bchan=156-170,172-186 dchan=171 # Global data loadzone= us defaultzone= us etc/asterisk/zapata.conf [channels] context=default hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 switchtype=euroisdn signalling=pri_cpe callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 group=2 callgroup=2 switchtype=qsig signalling=pri_net callerid=mycallerid immediate=no channel = 156-170 channel = 172-176 channel = 125-139 channel = 141-155 thanks and regards 2013/3/27 Yves A. yves...@gmx.de mailto:yves...@gmx.de Am 26.03.2013 17:57, schrieb Salaheddine Elharit: Hello, i have all the time this warning i use asterisk 1.4 all works without issue i don't have any problem (i can use the inbound and outbound calls without issue) i just want to know what is this WARNING thanks and regards WARNING[28151]: chan_zap.c:2404 pri_find_dchan: No D-channels available! Using Primary channel 140 as D-channel anyway! this can have different causes... mostly a wrong setting in your zaptel configuration file... this could be e.g. mixing american / european settings (e1/t1), wrong timing settings, wrong master / source clock setting, [...] post more details... what span (e1 or t1), which hardware, driver version, asterisk version, config files... regards, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http
Re: [asterisk-users] question about zapata.conf
it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about zapata.conf
hi, migrating from zaptel to dahdi HAS an impact... new config files, new options and a new channeldriver that has to be used in your dialplan ... you would have to select the DAHDI channel instead of your ZAP channel when dialing... if you´re to afraid to do it... then leave it as it is and follow the ntars-maxime (never touch a running system)... regards, yves Am 25.03.2013 16:15, schrieb Salaheddine Elharit: thank you so much fo the upgrade from zptel to dahdi, if there is any possibility to upgrade to dahdi without impacting my installation of asterisk and other application already installed in my server. if you can tell how to upgrade using dahdi drivers thanks and best regards 2013/3/25 Eric Wieling ewiel...@nyigc.com mailto:ewiel...@nyigc.com Service asterisk stop Service zaptel restart Service asterisk start -Original Message- From: asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine Elharit Sent: Monday, March 25, 2013 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] question about zapata.conf i use asterisk 1.4, how i can do to reload dirver 1.service asterisk stop 2 CLI reload chan_zap.so 3 service asterisk start that is right or i miss something ? 2013/3/25 Yves A. yves...@gmx.de mailto:yves...@gmx.de it depends a little bit on the driver and asterisk version... the safest way to become changes applied is to stop asterisk, reload the driver and than start asterisk again. regards, yves btw..: zaptel ist outdated... you should definitely upgrade using dahdi drivers... Am 25.03.2013 11:44, schrieb Salaheddine Elharit: hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . service zaptel restart or there is any other command Thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording with MixMonitor and AGI
hi, the music heard by MoH is configurable... so if you want silence... But hold could e.g. also be done by transferring a caller into a dynamic meetme room... yves Am 14.03.2013 08:43, schrieb Henrik Westerberg: Hi, The idea was to record an ongoing call by three party bridging on the mobile phone. Well my problem was to halt execution of the Dialplan so the server would not hang up the call. And I don´t want the server to say anything during the call. Now I solved this case as well by using Answer and then Record in the dialplan . So I´m not recording with MixMonitor. But just out of curiosity. How did you mean using hold (in answer/hold). Is that MusicOnHold? For me I can´t use that since I don´t want to make any noise. Is there another way? exten = 111,1,Answer() exten = 111,n,? I have tried using Wait with a long duration but have not succeeded to make it work as I want. I am using asterisk-java and originate calls to local channels. Regards, Henrik Från: Yves A. yves...@gmx.de mailto:yves...@gmx.de Datum: söndag 10 mars 2013 11:42 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com, Henrik Westerberg henrik.westerb...@ain.se mailto:henrik.westerb...@ain.se Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI Hi, so if your are ok with the way you solved part 1... alright, lets go to part 2.. but again... hu.. I don´t understand.. what do you mean with merging to a mobile phone? do you want do bridge the calls (three partys) or do you want to play the just recorded file from your server-initiated call into a another running call? what is by hand? the more explicit you are, the more helpful will be the answer. you ask but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up of course you can...you could e.g.: call into a queue call into a meetme room call with the help of a local channel into a context where you do nothing but answer / hold but as i said i did not quite catch what your objective really is... i just dont understand your scenario or cant imagine its sense. if you are a java programmer, i think your using the asterisk-java lib from s. reuter.. if so, you have any freedom, you could also use ami connection to listen to events to start and stop recordings and so on. regards, yves Am 09.03.2013 21:32, schrieb Henrik Westerberg: Hi, Thanks for your answer! 1. so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? Yes this is correct, and I prefer to do the transferring of the file to another server with my existing AGI. My AGIs are written in java. Today I the upload is done over http. Today I schedule the upload in the AGI script a couple of seconds after the channel is hang up. But the two lines might not be hung up at the same time. Your suggestion of always fixing the file is wise, it now seems to work fine after having been processed with sox. So now I think that this case 1 is ok for me :-) 2. and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? The idea is to perform a probe call with the only task of recording what the other party says. It will be merged by hand on a mobile phone to an ongoing call with another party. This could be done by calling out and letting AGI execute a RECORD FILE but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up Then I could easily use the MixMonitor and write the whole conversation in the dialplan with uploading similar to the first case. Any suggestions? Regards, Henrik Från: Yves A. yves...@gmx.de mailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Datum: torsdag 7 mars 2013 20:10 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI hi, hard to understand, what your objective is... at least for me ;-) so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? let me know, if i understood you right, the solution is not so hard to implement. In what language do you preferrably write your AGIs? (although
Re: [asterisk-users] Recording with MixMonitor and AGI
Hi, so if your are ok with the way you solved part 1... alright, lets go to part 2.. but again... hu.. I don´t understand.. what do you mean with merging to a mobile phone? do you want do bridge the calls (three partys) or do you want to play the just recorded file from your server-initiated call into a another running call? what is by hand? the more explicit you are, the more helpful will be the answer. you ask but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up of course you can...you could e.g.: call into a queue call into a meetme room call with the help of a local channel into a context where you do nothing but answer / hold but as i said i did not quite catch what your objective really is... i just dont understand your scenario or cant imagine its sense. if you are a java programmer, i think your using the asterisk-java lib from s. reuter.. if so, you have any freedom, you could also use ami connection to listen to events to start and stop recordings and so on. regards, yves Am 09.03.2013 21:32, schrieb Henrik Westerberg: Hi, Thanks for your answer! 1. so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? Yes this is correct, and I prefer to do the transferring of the file to another server with my existing AGI. My AGIs are written in java. Today I the upload is done over http. Today I schedule the upload in the AGI script a couple of seconds after the channel is hang up. But the two lines might not be hung up at the same time. Your suggestion of always fixing the file is wise, it now seems to work fine after having been processed with sox. So now I think that this case 1 is ok for me :-) 2. and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? The idea is to perform a probe call with the only task of recording what the other party says. It will be merged by hand on a mobile phone to an ongoing call with another party. This could be done by calling out and letting AGI execute a RECORD FILE but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up Then I could easily use the MixMonitor and write the whole conversation in the dialplan with uploading similar to the first case. Any suggestions? Regards, Henrik Från: Yves A. yves...@gmx.de mailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Datum: torsdag 7 mars 2013 20:10 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI hi, hard to understand, what your objective is... at least for me ;-) so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? let me know, if i understood you right, the solution is not so hard to implement. In what language do you preferrably write your AGIs? (although there is no absolute need for using an agi... you can all write down in your dialplan...) is there a special protocol requirement for saving/transferring the recorded voicefile (e.g. ftps)? One obstacle is, that the recorded file is not fully written _immediately_ after stopmixmonitor or hangup... this has to be taken care of and depending on your agi... it might be interrupted, if the call is hungup... but as you did not show your agi... these are just hints.. regards, yves Am 07.03.2013 16:21, schrieb Henrik Westerberg: Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten = s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten = _X.,1,NoOp(Will send call to ${EXTEN}) exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten = h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID}) exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is ${CC_CALLID}, CC_FILENAME is ${CC_FILENAME}) exten = _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e) If I want to make a recorded server callout from 0 to 08 I then originate a call via AMI to Local/0@outgoing-originate with context set to outgoing-originate-rec and extension to 08. The result will be something like
Re: [asterisk-users] Sending SMS from asterisk
Hi there, for sending SMSes I am using a 3G Modem and SMSlib... it is not bound to asterisk in any way, but I always wanted to integrate this possibility some day. I could not do so, because our landline provider (Vodafone) does not support it via E1 PRI lines... by I thought (never tried...) if it would be possible to use the SMS ServiceNumber from my mobile Provider...? I have a valid mobile contract, the number of the SMScc , my Cardnumber (t-mobile), my phonenumber and so on... so it should be possible, I think... but how? Has anybody a clue? regards, yves Am 09.03.2013 11:03, schrieb Miguel Oyarzo: Hi Bilal, It's not necessary to use a FXS port, you can compile install chan_dongle and buy a Huawei 3G dongle. We have running here a SMS solution with four 3G dongles, which sends over 20.000 SMS a month. In addition, I wrote an script able to send up to 12000 characters in concatenated SMS (the recipient receives a single SMS only) chan_dongle works very well. -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/9/2013 1:09 PM, Gerardo Barajas wrote: Yes, you can check solutions from sangoma and khomp. Saludos/Regards -- Ing. Gerardo Barajas Puente Proyectos Especiales/Preventa | www.neocenter.com http://www.neocenter.com T:+52 (55) 8590-9000 x 7003 On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW? From the other side, this is existed only in asterisk 1.8 or it is existed in asterisk 1.4? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording with MixMonitor and AGI
hi, hard to understand, what your objective is... at least for me ;-) so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? let me know, if i understood you right, the solution is not so hard to implement. In what language do you preferrably write your AGIs? (although there is no absolute need for using an agi... you can all write down in your dialplan...) is there a special protocol requirement for saving/transferring the recorded voicefile (e.g. ftps)? One obstacle is, that the recorded file is not fully written _immediately_ after stopmixmonitor or hangup... this has to be taken care of and depending on your agi... it might be interrupted, if the call is hungup... but as you did not show your agi... these are just hints.. regards, yves Am 07.03.2013 16:21, schrieb Henrik Westerberg: Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten = s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten = _X.,1,NoOp(Will send call to ${EXTEN}) exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten = h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID}) exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is ${CC_CALLID}, CC_FILENAME is ${CC_FILENAME}) exten = _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e) If I want to make a recorded server callout from 0 to 08 I then originate a call via AMI to Local/0@outgoing-originate with context set to outgoing-originate-rec and extension to 08. The result will be something like this: -- Executing [s@macro-ccdev2-rec:1] MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack == Begin MixMonitor Recording SIP/upps-ccm-tq01-003f -- Executing [h@outgoing-originate-rec:1] AGI(SIP/upps-ccm-tq01-003e, agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack -- SIP/upps-ccm-tq01-003eAGI Script agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed, returning 0 -- Executing [h@outgoing-originate-rec-dev2:1] AGI(SIP/upps-ccm-tq01-003f, agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack -- SIP/upps-ccm-tq01-003fAGI Script agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed, returning 0 == MixMonitor close filestream (mixed) == End MixMonitor Recording SIP/upps-ccm-tq01-003f Unfortunately I get two different calls to the h extension, but this I can cope with. The one without called is not interesting. The uploading will fail since the MixMonitor is still on when I try to upload the file. The file will not have a duration. It works when I schedule the uploading a while after from my agi application but I would rather not rely on a timeout. When I tried to run StopMixMonitor before the Agi call in the h extension, the first call fail and I never get any uploading with callid. -- Executing [s@macro-ccdev2-rec:1] MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new stack == Begin MixMonitor Recording SIP/upps-ccm-tq01-0043 -- Executing [h@outgoing-originate-rec-dev2:1] StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack == Spawn extension (outgoing-originate-rec-dev2, h, 1) exited non-zero on 'SIP/upps-ccm-tq01-0042' -- Executing [h@outgoing-originate-rec-dev2:1] StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack == MixMonitor close filestream (mixed) -- Executing [h@outgoing-originate-rec-dev2:2] AGI(SIP/upps-ccm-tq01-0043, agi://localhost/ajpbxtest.agi?path=uploadreccallid=) in new stack Am I missing something here? I also looked at the possibility to specify a command to execute when MixMonitor stops but I would rather handle the file uploading in my agi application. I also have another case: I want to dial out a call and record it. It will be a oneway-call from the server to a mobile. Do I need to get AGI-control of it and record with an AGI command or how can I hack it directly in the dial plan using MixMonitor? Best Regards, Henrik -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live
Re: [asterisk-users] Extension cant pickup calls but can transfer.
do you have only ONE phone, that can´t pickup, or is this a general problem? is pickup configured (feature.conf) AND enabled ? regards, yves Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension cant pickup calls but can transfer.
is it the same type and make of phone than one of the working ones? - compare (dtmf) settings, firmware release etc. - check call-group and pickup group... is the non working extension configured there? regards, yves Am 07.03.2013 20:28, schrieb Luis H. Forchesatto: Its only ONE phone who doesnt pickup calls. 2013/3/7 Yves A. yves...@gmx.de mailto:yves...@gmx.de do you have only ONE phone, that can´t pickup, or is this a general problem? is pickup configured (feature.conf) AND enabled ? regards, yves Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* * -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Att.* * Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com mailto:luis_forchesa...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension cant pickup calls but can transfer.
mmh... should work... (i think you checked double and applied any changes, right..? sometimes deleting the extension and configuring a new one can fulfil wonders...) I have no further tip... maybe elastix support or forum can help... if you are familiar with cli output and sip debugging... check cli output and sip debug output... good luck. yves Am 07.03.2013 20:38, schrieb Luis H. Forchesatto: Yes, both are configured in the same ata (linksys pap2) and the configuration options are the same. Call group and pick group are the same for both too. 2013/3/7 Yves A. yves...@gmx.de mailto:yves...@gmx.de is it the same type and make of phone than one of the working ones? - compare (dtmf) settings, firmware release etc. - check call-group and pickup group... is the non working extension configured there? regards, yves Am 07.03.2013 20:28, schrieb Luis H. Forchesatto: Its only ONE phone who doesnt pickup calls. 2013/3/7 Yves A. yves...@gmx.de mailto:yves...@gmx.de do you have only ONE phone, that can´t pickup, or is this a general problem? is pickup configured (feature.conf) AND enabled ? regards, yves Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* * -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Att.* * Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com mailto:luis_forchesa...@hotmail.com -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Att.* * Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com mailto:luis_forchesa...@hotmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] red alarm on span - do channels in the group automatically get skipped over?
hi, yes, this is the way, asterisk / the channeldriver handles it. you can simulate the failure of one span by just pulling out the cable and see what happens.. on top, you can influence the order, the channels are used by using dahdi/g1 or dahdi/G1... regards, yves Am 05.03.2013 07:31, schrieb Hose: Hello, If I put two spans' worth of channels, say 1-23 from span 1 and 25-47 in span 2, in one group, but only span 2 was showing OK and the other was down / showing a RED alarm, would asterisk automatically skip over trying to use channels 1-23 when doing outbound calls? e.g., dial(dahdi/g1/(number) would just jump to channel 25? Testing seems to bear this out, but I'm not positive about it. hose -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC and SQLIte3
hi, if you use realtime peers, and you want to see their states, you have to look in the database... if you want to see their states via cli, you have to set rtcachefriends=yes in your sip.conf... there are other settings that you might be interested in... : rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) rtsavesysname=yes ; Save systemname in realtime database at registration ; Default= no rtupdate=yes ; Send registry updates to database using realtime? (yes|no) ; If set to yes, when a SIP UA registers successfully, the ip address, ; the origination port, the registration period, and the username of ; the UA will be set to database via realtime. ; If not present, defaults to 'yes'. Note: realtime peers will ; probably not function across reloads in the way that you expect, if ; you turn this option off. rtautoclear=yes; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered? (yes|no|seconds) ; If set to yes, when the registration expires, the friend will ; vanish from the configuration until requested again. If set ; to an integer, friends expire within this number of seconds ; instead of the registration interval. ignoreregexpire=yes; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage regards, yves Am 17.02.2013 12:51, schrieb termo termosel: Hi, I had configured Asterisk to use default database located in /var/lib/asterisk/sqlite3dir/sqlite3.db. When I put odbc show in Asterisk's cli, It returns me that I have conected but when I put sip show peers,Asterisk doesn't found any peer or user. ubuntu*CLI odbc show ODBC DSN Settings - Name: asterisk DSN:asterisk-connector Last connection attempt: 1970-01-01 01:00:00 Pooled: No Connected: Yes ubuntu*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status Description Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] This mi configuration, /etc/odbci.ini [asterisk-connector] Description = SQLite3 database Driver = SQLite3 Database= /var/lib/asterisk/sqlite3dir/sqlite3.db /etc/odbcinst.ini [SQLite3] Description= SQLite3 ODBC Driver Driver=/usr/local/lib/libsqlite3odbc.so Setup=/usr/local/lib/libsqlite3odbc.so Threading=2 /etc/asterisk/extconfig.conf [settings] sipusers = odbc,asterisk,sip_buddies sippeers = odbc,asterisk,sip_buddies sipregs = odbc,asterisk,sip_buddies /etc/asterisk/func_odbc.conf [SQL] dsn=asterisk readsql=${ARG1} /etc/asterisk/modules.conf autoload=yes ;preload = res_odbc.so ;preload = res_config_odbc.so noload = pbx_gtkconsole.so ;load = pbx_gtkconsole.so noload = pbx_kdeconsole.so noload = app_intercom.so noload = chan_modem.so noload = chan_modem_aopen.so noload = chan_modem_bestdata.so noload = chan_modem_i4l.so noload = chan_capi.so load = res_musiconhold.so noload = chan_alsa.so ;noload = chan_oss.so noload = cdr_sqlite.so noload = app_directory_odbc.so ;noload = res_config_odbc.so ;noload = res_config_pgsql.so /etc/asterisk/res_odbc.conf [asterisk] enabled = yes dsn = asterisk-connector pre-connect = yes Can someone help me? Thanks, Jordi -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every
Re: [asterisk-users] ODBC and SQLIte3
looks like a mistake in your extconfig.conf... do you want to use realtime extensions too? for further instructions show us your extensions.conf and the verbose output of the cli showing the dialattempt... regards, yves Am 17.02.2013 14:31, schrieb termo termosel: Hi, I have add this options into Sip.conf but the CLI continues telling the same message: ubuntu*CLI sip show peers Name/username Host Dyn Forcerport ACL Port Status Description Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] I have two users in my slite3.db. but Asterisk doesn't show me. It is how asterisk can't access into this database. When I go to call, Asterisk tells me that extension xxx is not found in phones context. Thanks, Jordi Date: Sun, 17 Feb 2013 13:00:44 +0100 From: yves...@gmx.de To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ODBC and SQLIte3 hi, if you use realtime peers, and you want to see their states, you have to look in the database... if you want to see their states via cli, you have to set rtcachefriends=yes in your sip.conf... there are other settings that you might be interested in... : rtcachefriends=yes ; Cache realtime friends by adding them to the internal list ; just like friends added from the config file only on a ; as-needed basis? (yes|no) rtsavesysname=yes ; Save systemname in realtime database at registration ; Default= no rtupdate=yes ; Send registry updates to database using realtime? (yes|no) ; If set to yes, when a SIP UA registers successfully, the ip address, ; the origination port, the registration period, and the username of ; the UA will be set to database via realtime. ; If not present, defaults to 'yes'. Note: realtime peers will ; probably not function across reloads in the way that you expect, if ; you turn this option off. rtautoclear=yes; Auto-Expire friends created on the fly on the same schedule ; as if it had just registered? (yes|no|seconds) ; If set to yes, when the registration expires, the friend will ; vanish from the configuration until requested again. If set ; to an integer, friends expire within this number of seconds ; instead of the registration interval. ignoreregexpire=yes; Enabling this setting has two functions: ; ; For non-realtime peers, when their registration expires, the ; information will _not_ be removed from memory or the Asterisk database ; if you attempt to place a call to the peer, the existing information ; will be used in spite of it having expired ; ; For realtime peers, when the peer is retrieved from realtime storage, ; the registration information will be used regardless of whether ; it has expired or not; if it expires while the realtime peer ; is still in memory (due to caching or other reasons), the ; information will not be removed from realtime storage regards, yves Am 17.02.2013 12:51, schrieb termo termosel: Hi, I had configured Asterisk to use default database located in /var/lib/asterisk/sqlite3dir/sqlite3.db. When I put odbc show in Asterisk's cli, It returns me that I have conected but when I put sip show peers,Asterisk doesn't found any peer or user. ubuntu*CLI odbc show ODBC DSN Settings - Name: asterisk DSN:asterisk-connector Last connection attempt: 1970-01-01 01:00:00 Pooled: No Connected: Yes ubuntu*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status Description Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] This mi configuration, /etc/odbci.ini [asterisk-connector] Description = SQLite3 database Driver = SQLite3 Database= /var/lib/asterisk/sqlite3dir/sqlite3.db /etc/odbcinst.ini [SQLite3] Description= SQLite3 ODBC Driver Driver=/usr/local/lib/libsqlite3odbc.so Setup=/usr/local/lib/libsqlite3odbc.so Threading=2
Re: [asterisk-users] cisco 7940 and asterisk 11
maybe not related to your problem, but I had a similar effect after upgrading my 1.6 to 11. not on phones (pbx was used as ivr only), but voicefile were played in a robo style unconditionally. this effect could only be gotten rid of by rebooting the server. i had to completely clean the installation and rebuild 11 from scratch... i think some mp3 classes have caused the effect... (although no mp3 files were used..) when the effect occurred, there was nothing that hinted to any problem... cpu usage, network etc. everything was fine, but even only restarting asterisk did not help... i think some libs got messed up during update. so i´d recommend to rebuild the server complete from scratch... if the problem still exists after new build... it would be interesting to know, if you just took the same config-files from your previous version which would maybe cause problems... if all fails, i would then go deeper into network analysis and trace the traffic. meanwhile i administer around 10 asterisk boxes and i always use ubuntu 12.04 lts and latest asterisk 11 on dell r3/4/6xx servers... up to now everything runs fine.. regards, yves Am 14.02.2013 07:20, schrieb Julian Lyndon-Smith: very polite *bump* this is a real issue for us - anyone got _any_ clues or ideas ? Thanks ;) On 12 February 2013 14:29, Julian Lyndon-Smith aster...@dotr.com mailto:aster...@dotr.com wrote: Ever since we upgraded to asterisk 11 we have had audio problems with our cisco 7940 phones. The problems manifest themselves by the conversation turning robotic or into silence (to the extent our agents are saying hello? hello? and the customer is saying I hear you just fine We had to change pedantic=no in sip.conf to allow the phones to register We are assuming that it is the phone=asterisk combination because a) the call recordings of the conversation are perfect (no noise on the line, conversation is clear) but it is apparent that the agent cannot hear the customer sometimes (Hello?) b) we have replaced the cables and switches between the phones and the pbx c) we don't have the same problem with Aastra 9133i or Polycom 331 phones Are there any settings in sip.conf that may help this , or a particular firmware ? Are there any known audio problems with cisco 7940 and asterisk 11 ? Many thanks Julian -- Julian Lyndon-Smith IT Director, Dot R Limited I don't care if it works on your machine! We are not shipping your machine! The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- Julian Lyndon-Smith IT Director, Dot R Limited I don't care if it works on your machine! We are not shipping your machine! The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variables set by AGI lost in dialplan
are the calls always handled in the same manner, or sometimes different, e.g. with conferencing, bridging, transferring, using local channels and and and...? in that case try to use two underscores before the variablename to use inheritance. maybe you accidentially swap the channels? you could try to set the variables with help of the shared function to set it for both channels... regards, yves Am 14.02.2013 10:40, schrieb Deepesh D: Hello, I am using asterisk 1.8.17.0 with a fast agi written in C The following is a part of my dialplan exten = _X.,n,MSet(my_var=0,my_var1=0) exten = _X.,n,AGI ;; Call to a fast agi to set values of my_var my_var1 exten = _X.,n,Log(NOTICE,${my_var} ${my_var1}) ;; log the values to asterisk messages Inside the AGI I do some calculations and set the values of my_var and my_var1 variables like SET VARIABLE my_var 0.008 SET VARIABLE my_var1 0.009 The problem I am facing is that sometimes the variables are wrongly received as 0 (zero) in the dialplan even if the AGI has set it to a non-zero value. Inside the AGI I am logging the values of variables to a log file, and the log file always shows non-zero values. But in my asterisk messages file the values are zero for some calls. This error does not happen for all calls and is not reproducible, it is random. My asterisk server handles about 100 calls per minute, so its impossible for me to do an 'agi set debug' and observer the output -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Realtime Extension... strange behaviour
Hi, I encountered a strange behaviour using realtime extensions... (on Asterisk 11.2) when I use the following static dialplan, everything works as expected..: [from-sip] exten = 110,1,Dial(DAHDI/g0/${EXTEN}) exten = 112,1,Dial(DAHDI/g0/${EXTEN}) exten = _XXX,1,Dial(SIP/${EXTEN}) exten = _X.,1,Dial(DAHDI/g0/${EXTEN}) will say... if a sip phone calls 110 or 112 the call is routed into PSTN (german emergency call) if a sip phone calls any three digit number, the call should be routet to the corresponding SIP user and if a sip phone calls any other number the call should be routed into PSTN... thats ok and works as expected. when I change to realtime: [from-sip] switch = Realtime and put the diaplan into the database idcontextextenpriorityappappdata 1from-sip1101DialDAHDI/g0/${EXTEN} 2from-sip1121DialDAHDI/g0/${EXTEN} 3from-sip_XXX1DialSIP/${EXTEN} 4from-sip_X.1DialDAHDI/g0/${EXTEN} only the emergency calls work and any other call goes to DAHDI... I cant reach any other SIP phone. Even when swapping the content of the rows 3 and 4 in the database to idcontextextenpriorityappappdata 1from-sip1101DialDAHDI/g0/${EXTEN} 2from-sip1121DialDAHDI/g0/${EXTEN} 3from-sip_X.1DialDAHDI/g0/${EXTEN} 4from-sip_XXX1DialSIP/${EXTEN} makes no difference... I thought, using realtime extensions would read the dialplan from top to bottom, ordered by id... but it seems to be ignored somehow and the extension _X. catches the calls before the extensionpattern _XXX is reached. I _could_ avoid this be prefixing external numbers with a leading 0 for example... but I dont want to... as I said.. using static extension via extensions.conf the dialplan works as expected... Am I missing something? regards, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] About Asterisk with Digium TE405P PRI ISDN cad
Hi, I think, you mean connecting the two boxes directly with a cable... not via PSTN, right? 1.) You need a special cross-over cable to connect one Port directly to another Port... (if you want to crimp it yourself, you can find the associated Pins via Google... ethernet crossover cables do not work as they have different links) 2.) configure one end as master (CPN) and the other asterisk as Network (CPN), otherwise you´ll get timing issues... thats all... regards, yves Am 11.02.2013 14:00, schrieb Shitian Long: Hello, I am trying to connect two asterisks with PRI connection. One asterisk has TE405P Quad PRI ports card, anther asterisk has TE110P 1 PRI port card. I am wondering if there would be some step by step guide that I could follow to to this kind of connection? Thanks -- from longst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] access control softphone registration through asterisk
Hi, are you using realtime extensions or the classic config-file extension.conf ? One way to go yould be to implement the allowed / not allowed logic in the context of your sip users. check their permissions and if they are allowed to call... continue with the dialplan, if not, route them to a voiceprompt saying that the call is prohibited due to whatever reasons... To do so, take a look at the dialplan functions if and db. Of course you somehow have to set a flag in asterisk, that decides about permissions... Don´t know which way you will programmatically set or clear this flag... there are hundreds of possibilites... the easiest way I think would be to use the asterisk build-in database (therefore the hint to the function db...) regards, yves Am 08.02.2013 22:18, schrieb Muhammad: Hi, I wana control my SIP register from asterisk. I other hand, when users login into their softphone, dont access to call and when I give them access, they can call. I dont know it's right way to plan my scenario/? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] special conference room
barat and danny, thank you for your input... I am using asterisk 11.2 and i read about meetme. Yes, it has many switches and options and can help me a lot... but as you already said... does _almost_ all features.. unfortunately I need ALL the constraints fulfilled... therefore i admit I have not tried it in deep, because just from reading the doc I realized, that it wont fit all my needs... btw.: I understood the mute switch to disable the callers to talk to the conference.. (so to say it mutes the callers microphone, not his earphones am I wrong? nevertheless... any more hints for my original feature-request? thank you all, yves Am 16.01.2013 19:03, schrieb Bharat Lalcheta: Please study meetme application's options. You will get almost all feature you ask for in it On Jan 16, 2013 5:37 AM, Yves A. yves...@gmx.de mailto:yves...@gmx.de wrote: Hi list, I am in need of a special asterisk conference room with the following constraints: - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how to realize that are highly appreciated.. Thanx in advance, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] special conference room
ok, now i have got some very valuable information to start off with. thank you all. i´ll be back to report success or further questions... just one thing, that i think might be a showstopper that i may have not explained clear enough...: muting and unmuting a caller should have the effect, that the caller can talk to the moderator or not... any caller should NEVER hear what other callers are talking... may he be muted or not... yves Am 16.01.2013 23:01, schrieb Danny Nicholas: From what I read, neither confbridge or meetme have the whisper feature built-in; This doesn't matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web interface. Let's say Yves' special conference is . The moderator would start using this command Exten = s,1,meetme() The participants would do Exten = s,1,meetme(,m) -- muted so they can listen but not talk - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator The moderator would need to be able to enumerate the conference by doing Asterisk --rx core show channels verbose|grep meetme This is supposed to be doable from the dialplan but my google-fu failed me on it. - the moderator must be able to mute and unmute any caller at any time Establish a maximum number of users and set this up for each one Exten = 99,1,meetmeadmin(,M,1) let user 1 talk Exten = 199,1,meetmeadmin(,m,1) turn user 1 back off - the moderator must be able to talk to all callers or to a specific caller. Exten = 901,1,chanspy(SIP/XXX,w) - the modetator must be able to kick off any caller at any time... Exten = 299,1,meetmeadmin(,k,1) kick out user 1 Exten = 666,1,meetmeadmin(,K) shut it down *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Don Kelly *Sent:* Wednesday, January 16, 2013 3:34 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] special conference room Sounds like a conference with all attendees permanently muted (except the moderator). The moderator uses whisper to communicate with individuals. --Don *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] mailto:[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Yves A. *Sent:* Wednesday, January 16, 2013 3:11 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] special conference room barat and danny, thank you for your input... I am using asterisk 11.2 and i read about meetme. Yes, it has many switches and options and can help me a lot... but as you already said... does _almost_ all features.. unfortunately I need ALL the constraints fulfilled... therefore i admit I have not tried it in deep, because just from reading the doc I realized, that it wont fit all my needs... btw.: I understood the mute switch to disable the callers to talk to the conference.. (so to say it mutes the callers microphone, not his earphones am I wrong? nevertheless... any more hints for my original feature-request? thank you all, yves Am 16.01.2013 19:03, schrieb Bharat Lalcheta: Please study meetme application's options. You will get almost all feature you ask for in it On Jan 16, 2013 5:37 AM, Yves A. yves...@gmx.de mailto:yves...@gmx.de wrote: Hi list, I am in need of a special asterisk conference room with the following constraints: - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how to realize that are highly appreciated.. Thanx in advance, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
[asterisk-users] special conference room
Hi list, I am in need of a special asterisk conference room with the following constraints: - there is one admin / moderator and several normal callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how to realize that are highly appreciated.. Thanx in advance, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk console suddenly extremely verbose...
Hi there. I started the console today to reload the extensions.conf file ; only to be greeted with extremely verbose console. Seems related to the zaptel card: Example: Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 020 P/F: 1 0 bytes of data voip*CLI [ 00 01 01 2f ] voip*CLI Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 023 P/F: 1 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 22 to (but not including) 23 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 timer This is repeating every 10s or so... Any ideas what this message means and is there a way to prevent it from happening. No changes has been made on this asterisk box in years (running old 1.4.25 if it ain't broken version) Thanks in advance JY -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk console suddenly extremely verbose...
Hi On 16 December 2011 13:24, Richard Mudgett rmudg...@digium.com wrote: You have pri intense debug span x enabled. Disable with pri no debug span x. Thanks... I couldn't find any configuration file showing this ; but ran the command in the CLI... Seems to have done it. I really wonder how it could have been turned on ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redirecting a Channel more than three times...
Hi folks, could someone please try to confirm the following (mis)behaviour of my asterisk? Imagine the following scenario: Caller A calls the central. Central picks up, talks to Caller A which wants to be connected to employee X. Central puts Caller A on hold by Redirecting the Channel to a Queue. Central calls emplyee X and bridges both channels... everybody is happy. But..: Caller A calls the central. Central picks up, talks to Caller A which wants to be connected to employee X. Central puts Caller A on hold by Redirecting the Channel to a Queue. Central calls emplyee X and X doesn´t want to talk with Caller A Central and employee hang up.. Central pulls Caller A back from Queue (again, with Redirecting the channel to its own extension) Caller A now want to talk with employee Y and so on This game works exactly three times... when the central wants to pull back the Caller from the Queue for the third time, the call is hungup. I searched and searched, but could not find anything about a redirect-limit or so... what, if there is no such limit, am I doing wrong? If there is such a limit.. where is it configured? thank you anyways, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Redirecting a Channel more than three times...
Hi Danny, I decided against Parking Calls, because it seemed quite complicated and useless for me... as far as i remember, parkedcalls return automagically after a timeout which was not desirable. I would have to rewrite a lot of code, if i have to change... but there must be a reason for this misbehaviour, and i think its hardcoded in the asterisk-source. somewhere seems to be a counter that counts the redirects... it maybe useful in some case, maybe to avoid loops or something similar to bounces in emails, but in my case its undesired... because i am using trixbox / freepbx the dialplan is very complicated, but it showed me no hint of beeing responsible for this... the cli-output gives no hint. yves Am 24.09.2010 15:10, schrieb Danny Nicholas: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Friday, September 24, 2010 6:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Redirecting a Channel more than three times... Hi folks, could someone please try to confirm the following (mis)behaviour of my asterisk? Imagine the following scenario: Caller A calls the central. Central picks up, talks to Caller A which wants to be connected to employee X. Central puts Caller A on hold by Redirecting the Channel to a Queue. Central calls emplyee X and bridges both channels... everybody is happy. But..: Caller A calls the central. Central picks up, talks to Caller A which wants to be connected to employee X. Central puts Caller A on hold by Redirecting the Channel to a Queue. Central calls emplyee X and X doesn´t want to talk with Caller A Central and employee hang up.. Central pulls Caller A back from Queue (again, with Redirecting the channel to its own extension) Caller A now want to talk with employee Y and so on This game works exactly three times... when the central wants to pull back the Caller from the Queue for the third time, the call is hungup. I searched and searched, but could not find anything about a redirect-limit or so... what, if there is no such limit, am I doing wrong? If there is such a limit.. where is it configured? thank you anyways, yves #1. Have you looked at the CLI output for this scenario #2. Why don't you use Parking instead of queue? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - SIP-ROUTER - Internet = no audio
thanks brian, yes, i am aware that sip is only responsible for signalling and therefor my conclusion was, that it has got something to do with nat / firewall / the router... meanwhile i´ve got it solved... although the sip-provider tried to convince me, that the misconfiguration is on my asterisks´ side, i penetrated the support until they looked over it again and... what should i say... finally they had to admit, that the router had a wrong acesslist. they corrected it and now it works. yves Brian schrieb: On Fri, 2010-02-12 at 02:18 +0100, Yves Arikoglu wrote: Hi, I am breaking my fingers in configuring an asterisk (1.6) to successfully transmit audio with the following setup: asterisk, resides in local network, ip is 10.26.208.252 versatel business router (directly connected to a dsl, configured by sip-provider), WAN ip 89.244.13.25 versatel sip-proxy ip 89.244.13.10 in sip.conf I have: [general] bindaddr=0.0.0.0 externip=89.244.13.25 localnet=10.26.208.0/255.255.252.0 nat=yes qualify=yes the local sip phones register correctly and can make calls between each other with audio. the local sip phones CAN make outbound calls via the sip-provider... will say, destination phone rings, but there is no audio (on both legs) after pickup... external phones can call my sip-number... the call comes into the asterisk, the sip-extension rings, but after pickup... no audio at all. even if i route the call from external to a queue or something else... i see, that asterisk is playing voicefiles, but the caller does not hear anything. because sip-signalling works in any ways, but audio not, i think its got something to do with nat... but there is no firewall between asterisk and the router or between the router and the internetconnection from versatel... and i already tried millions of combinations of using nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m stuck as i was never ever stuck before :-( any hints? anybody? You are aware that SIP only sets up, monitors and takes the call down? The audio stream is RDP and on higher ports. My guess is that the audio stream on inbound calls is not arriving where it should be - or is blocked. This could be router or nat, but one thing jumps out to me: Does your Asterisk Server itself have something set up in the built in iptables firewall blocking udp inbound traffic in the port range 15000:2? The output of the command 'iptables -nvL' will tell you pretty quickly. HTH. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - SIP-ROUTER - Internet = no audio
Hi, I am breaking my fingers in configuring an asterisk (1.6) to successfully transmit audio with the following setup: asterisk, resides in local network, ip is 10.26.208.252 versatel business router (directly connected to a dsl, configured by sip-provider), WAN ip 89.244.13.25 versatel sip-proxy ip 89.244.13.10 in sip.conf I have: [general] bindaddr=0.0.0.0 externip=89.244.13.25 localnet=10.26.208.0/255.255.252.0 nat=yes qualify=yes the local sip phones register correctly and can make calls between each other with audio. the local sip phones CAN make outbound calls via the sip-provider... will say, destination phone rings, but there is no audio (on both legs) after pickup... external phones can call my sip-number... the call comes into the asterisk, the sip-extension rings, but after pickup... no audio at all. even if i route the call from external to a queue or something else... i see, that asterisk is playing voicefiles, but the caller does not hear anything. because sip-signalling works in any ways, but audio not, i think its got something to do with nat... but there is no firewall between asterisk and the router or between the router and the internetconnection from versatel... and i already tried millions of combinations of using nat=yes/no/route, qualify=yes/no, canreinvite=yes/no and and and and i´m stuck as i was never ever stuck before :-( any hints? anybody? thanks, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd error mssage on DAHDI lines
hi, you can lookup the causes in the sources check you dahdi-configuration (especially the groups...) is there everything ok? what does dahdi_tools or the other cli-commands say, that give you information about the available channels? yves /* Causes for disconnection (from Q.931) */ #define AST_CAUSE_UNALLOCATED1 #define AST_CAUSE_NO_ROUTE_TRANSIT_NET 2 #define AST_CAUSE_NO_ROUTE_DESTINATION 3 #define AST_CAUSE_CHANNEL_UNACCEPTABLE 6 #define AST_CAUSE_CALL_AWARDED_DELIVERED7 #define AST_CAUSE_NORMAL_CLEARING16 #define AST_CAUSE_USER_BUSY17 #define AST_CAUSE_NO_USER_RESPONSE18 #define AST_CAUSE_NO_ANSWER19 #define AST_CAUSE_CALL_REJECTED21 #define AST_CAUSE_NUMBER_CHANGED22 #define AST_CAUSE_DESTINATION_OUT_OF_ORDER 27 #define AST_CAUSE_INVALID_NUMBER_FORMAT 28 #define AST_CAUSE_FACILITY_REJECTED29 #define AST_CAUSE_RESPONSE_TO_STATUS_ENQUIRY30 #define AST_CAUSE_NORMAL_UNSPECIFIED31 #define AST_CAUSE_NORMAL_CIRCUIT_CONGESTION 34 #define AST_CAUSE_NETWORK_OUT_OF_ORDER 38 #define AST_CAUSE_NORMAL_TEMPORARY_FAILURE 41 #define AST_CAUSE_SWITCH_CONGESTION42 #define AST_CAUSE_ACCESS_INFO_DISCARDED 43 #define AST_CAUSE_REQUESTED_CHAN_UNAVAIL44 #define AST_CAUSE_PRE_EMPTED45 #define AST_CAUSE_FACILITY_NOT_SUBSCRIBED 50 #define AST_CAUSE_OUTGOING_CALL_BARRED 52 #define AST_CAUSE_INCOMING_CALL_BARRED 54 #define AST_CAUSE_BEARERCAPABILITY_NOTAUTH 57 #define AST_CAUSE_BEARERCAPABILITY_NOTAVAIL 58 #define AST_CAUSE_BEARERCAPABILITY_NOTIMPL 65 #define AST_CAUSE_CHAN_NOT_IMPLEMENTED 66 #define AST_CAUSE_FACILITY_NOT_IMPLEMENTED 69 #define AST_CAUSE_INVALID_CALL_REFERENCE81 #define AST_CAUSE_INCOMPATIBLE_DESTINATION 88 #define AST_CAUSE_INVALID_MSG_UNSPECIFIED 95 #define AST_CAUSE_MANDATORY_IE_MISSING 96 #define AST_CAUSE_MESSAGE_TYPE_NONEXIST 97 #define AST_CAUSE_WRONG_MESSAGE 98 #define AST_CAUSE_IE_NONEXIST99 #define AST_CAUSE_INVALID_IE_CONTENTS 100 #define AST_CAUSE_WRONG_CALL_STATE 101 #define AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE 102 #define AST_CAUSE_MANDATORY_IE_LENGTH_ERROR 103 #define AST_CAUSE_PROTOCOL_ERROR111 #define AST_CAUSE_INTERWORKING127 Richard Kenner schrieb: What's this: -- Attempting call on DAHDI/g1/9removed for application Wait(5) (Retry 1) -- Requested transfer capability: 0x00 - SPEECH -- Channel 0/2, span 1 got hangup, cause 44 -- Forcing restart of channel 0/2 on span 1 since channel reported in use -- Hungup 'DAHDI/2-1' Where can I look up cause 44. And if this is the sort of transient error that seems to be implied by the Forcing restart message, why isn't it retried? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops
do you use the qualify=yes option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip.conf with versatel and two NICs very strange problem
Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear anything. So i am quite sure that is has something to do with firewalls, natting and so on but i?ve read hundreds of pages and tried thousands of setting but i cant get audio to work.. the strange thing is... when i call the versatel-sip-number from my mobile phone, i see the call coming in in the cli, i see the voiceprompts that asterisk plays, but even there I cant hear anything on my mobile. next strange thing: i defined 2 sip-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... no audio. but when one sip extension calls a wrong number... the cannot be completed message is hearable. i configured a queue with moh and even this works... but why cant to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and interface is chosen for audio when a call comes in? thanks, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem
thanks, i tried this already but unfortunately no change. any further suggestions or answers concerning my other questions? thanx, yves Cary Fitch schrieb: As a guess, they can both talk to the server, but can't talk to each other. What is common to that is they may be trying to reinvite each other, and there is no path through the respective routers/firewalls to the other. So if reinvite is set to yes, set it to no, in both phone profiles on the server. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu Sent: Monday, January 25, 2010 7:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear anything. So i am quite sure that is has something to do with firewalls, natting and so on but i?ve read hundreds of pages and tried thousands of setting but i cant get audio to work.. the strange thing is... when i call the versatel-sip-number from my mobile phone, i see the call coming in in the cli, i see the voiceprompts that asterisk plays, but even there I cant hear anything on my mobile. next strange thing: i defined 2 sip-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... no audio. but when one sip extension calls a wrong number... the cannot be completed message is hearable. i configured a queue with moh and even this works... but why cant to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and interface is chosen for audio when a call comes in? thanks, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf with versatel and two NICs very strange problem
thanx... a typo... the routers local ip is 10.26.208.253 yves Tim Nelson schrieb: - Yves Arikoglu yves...@gmx.de wrote: Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y Either a typo or you have an IP conflict? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF reception during WaitForSilence
Hello, I wrote a little AGI-Script that implements an IVR (using asterisk 1.6). The whole conversation is recorded and at some points the caller should tell some information. I detect the silence (WaitForSilence) to go to the next step in the IVR. Until now everything is OK, but... some information the user gives (or speaks) is numeric... some users have the habit, to enter numeric information via the phonekeypad (ergo creating dtmf-tones) but I cant process DTMF-Input during WaitForSilence. How can I achive that both works simultaneously? I mean recording the spoken digits AND detecting DTMF-Input AND detecting silence to know, when Input has finished... (I want to avoid that users have to finish their input with the pound-key...) ? Btw.: why are the DTMF-Tones, that a user enters, not hearable in the recording? Thanks for your help and hints, Yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel - DAHDI: now echo
Hello I have upgraded our asterisk box from zaptel to dhadi two weeks ago... Since, there has been quite a significant amount of echo when making a call. Only for the local outgoing call, the person on the other side doesn't hear any echo. This is with a TE-110P ISDN PRI card .. I've pretty much took the original zaptel configuration and used it as-is with the dahdi one ; to no available.. Any help would be greatly appreciated. Here is what zaptel.conf and zapata.conf used to be: /etc/zaptel.conf: loadzone = au defaultzone=au #TE110P span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=16 /etc/asterisk/zapata.conf [channels] language=en usecallerid=yes hidecallerid=no callerid=asreceived restrictcid=no usecallingpres=yes ; ISDN Exchange Lines (Fractional E1 PRA10) switchtype=euroisdn signalling=pri_cpe immediate=no pridialplan=unknown prilocaldialplan=unknown overlapdial=yes echocancel=yes echocancelwhenbridged=yes echotraining=256 rxgain=1.0 txgain=8.0 context=incoming faxdetect=incoming group=1 channel=1-10 now for the dahdi configuration: /etc/dahdi/system.conf: loadzone = au defaultzone=au #TE110P span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=16 /etc/asterisk/chan_dahdi.conf [channels] language=en usecallerid=yes hidecallerid=no callerid=asreceived restrictcid=no usecallingpres=yes switchtype=euroisdn signalling=pri_cpe immediate=no pridialplan=unknown prilocaldialplan=unknown echocancel=yes echocancelwhenbridged=yes echotraining=256 rxgain=1.0 txgain=8.0 context=incoming faxdetect=incoming group=1 channel=1-10 --- From reading the various documentation, I was convinced that moving from zaptel to dahdi was almost just a matter of renaming the configuration file... Am I mistaken ? Thank you in advance for any help. Jean-Yves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel - DAHDI: now echo
Hi That was a fast answer, impressive ! 2009/8/18 Kevin P. Fleming kpflem...@digium.com: Did you read the upgrade documentation that comes with DAHDI, specifically from UPGRADE.txt: I did, but I guess I did not pay enough attention... * It is no longer possible to select a software echo canceler at compile time to build into dahdi.ko; all four included echo cancelers (MG2, KB1, SEC and SEC2) are built as loadable modules, and if the Digium HPEC binary object file has been placed into the proper directory the HPEC module will be built as well. Any or all of these modules can be loaded at the same time, and the echo canceler to be used on the system's channels can be configured using the dahdi_cfg tool from the dahdi-tools package. Note: It is *mandatory* to configure an echo canceler for the system's channels using dahdi_cfg unless the interface cards in use have echo canceler modules available and enabled. There is *no* default software echo canceler with DAHDI. So, knowing my card (a Digium TE-110P, which AFAIK doesn't have any hardware echo cancellation module)... Which software echo canceller should I be using ? Is see that there are particular software configuration available , but I haven't had a clue on what they are for, nor did I find documentation about it... I'm not building asterisk nor dahdi myself, but instead rely on packaged from ATrpms.conf Thank you Jean-Yves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel - DAHDI: now echo
Hi 2009/8/18 Tzafrir Cohen tzafrir.co...@xorcom.com: Something is missing here... http://docs.tzafrir.org.il/dahdi-tools/#_echo_canceller_modules Thanks .. I added to /etc/dahdi/system.conf the following: echocanceller=mg2,1-10 However, I have no clue about the various echo canceller, between mg2, kb1, sec2, and sec which one will provide the best performance ? (knowing that I was happy with whatever zaptel was doing before) JY ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Read Command
You first use the Read application : exten = s,n,Read(ANS|filetoplay) And then use GotoIfs by checking the ${ANS} variable to do the logic (re-ask if bad response, else continue in dialplan). On Sun, 2008-08-24 at 23:10 -0700, Joe Carroll wrote: I’ve search the world over…. but I haven’t figured out a way to have valid/invalid options for entry when using the Read command… I need to set a variable, but only want to allow certain values to be valid options for that variable… Any ideas? Thanks in advance.. -JC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR question
You could use func_odbc in your dialplan, check here : http://www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc Yves. On Thu, 2008-08-21 at 14:57 +0300, Szasz Szabolcs wrote: Hi! I'm setting up my IVR system, how can I register in a mysql database the IVR menus accessed by the clients ? Thanks a lot, Szasz Szabolcs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR question
Sorry, maybe I misunderstood your question. If you want the dialplan to be in a MySQL dabtase, check here : http://www.voip-info.org/wiki/view/Asterisk+configuration+from+database Works great, but the documentation is sometimes a bit outdated. Good luck. Yves. On Thu, 2008-08-21 at 14:57 +0300, Szasz Szabolcs wrote: Hi! I'm setting up my IVR system, how can I register in a mysql database the IVR menus accessed by the clients ? Thanks a lot, Szasz Szabolcs ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playback don't play the beginning if a sound file
Hello, I'm using this dialplan to let user record messages. The recording part works quite fine, but there is something strange : When Asterisk plays vm-torerecord, it misses the beginning, I only hear the few last seconds (vm-torerecord is a sound file that was in the asterisk-sounds cvs repo, but I simply renamed it). I've looked on voip-info.org, googled anything I could think about and checked on bugs.digium.com, I don't have any clue of what's going on. Does anyone has an idea ? Thanks. Here is my dialplan : [record] exten = s,1,Answer exten = s,n,Set(counter=1) exten = s,n,NoOp(${counter}) exten = s,n,GotoIf($[${counter} = 1]?record) exten = s,n(next),System(/bin/rm -f /var/lib/asterisk/sounds/${RECORDED_FILE}.wav) exten = s,n(record),Set(counter=$[${counter}+1]); exten = s,n,GotoIf($[${counter} 3]?i,1) exten = s,n,Playback(vm-intro) exten = s,n,Record(webrecord%d:wav,10,60) exten = s,n,Wait(1) exten = s,n,Set(CDR(userfield)=${RECORDED_FILE}) exten = s,n,Playback(${RECORDED_FILE}) exten = s,n(askretry),Background(vm-torerecord) exten = s,n,WaitExten(5) exten = i,1,Goto(s,askretry) exten = 3,1,Goto(s,next) exten = t,1,Set(CDR(userfield)=${RECORDED_FILE}) exten = t,n,Hangup ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback don't play the beginning if a sound file
It seems this has something to do with the Wait() before the Playback (Background behaves the same). If I remove the Wait, the next Playback is just fine, otherwise it truncates the beginning of the message. On Mon, 2008-05-05 at 10:41 +0200, Yves Räber wrote: Hello, I'm using this dialplan to let user record messages. The recording part works quite fine, but there is something strange : When Asterisk plays vm-torerecord, it misses the beginning, I only hear the few last seconds (vm-torerecord is a sound file that was in the asterisk-sounds cvs repo, but I simply renamed it). I've looked on voip-info.org, googled anything I could think about and checked on bugs.digium.com, I don't have any clue of what's going on. Does anyone has an idea ? Thanks. Here is my dialplan : [record] exten = s,1,Answer exten = s,n,Set(counter=1) exten = s,n,NoOp(${counter}) exten = s,n,GotoIf($[${counter} = 1]?record) exten = s,n(next),System(/bin/rm -f /var/lib/asterisk/sounds/${RECORDED_FILE}.wav) exten = s,n(record),Set(counter=$[${counter}+1]); exten = s,n,GotoIf($[${counter} 3]?i,1) exten = s,n,Playback(vm-intro) exten = s,n,Record(webrecord%d:wav,10,60) exten = s,n,Wait(1) exten = s,n,Set(CDR(userfield)=${RECORDED_FILE}) exten = s,n,Playback(${RECORDED_FILE}) exten = s,n(askretry),Background(vm-torerecord) exten = s,n,WaitExten(5) exten = i,1,Goto(s,askretry) exten = 3,1,Goto(s,next) exten = t,1,Set(CDR(userfield)=${RECORDED_FILE}) exten = t,n,Hangup ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto in Realtime extensions
That's very unfortunate. I use now a workaround : I'm just switching (with gotos) between extensions and use some macros but always within the same context. I'll try to remeber it for next time :) Cheers, Yves. On Fri, 2008-02-08 at 14:36 +0200, Atis Lezdins wrote: On 2/8/08, Yves Räber [EMAIL PROTECTED] wrote: * Version: Asterisk 1.4.14 * Commas instead of pipes = already tried, this is not working at all * Realtime switch for script_13_0 = No, should I ? This would be really a shame, I want to use realtime BECAUSE I don't want to play with my extensions.conf file. (I'm building a web interface that has to generate the contexts). Yes, unfortuneately that's the thing you have to do. You have to add each context you want - in static conf file like this: [db_na] switch = Realtime/db_na [db_busy] switch = Realtime/db_busy You can have as many extensions you like with whatever commands, but contexts still should be registered. Generally editing and debugging of complete dialplan in DB is not so easy - so you should keep your main logic in static, but use realtime for data that actually changes. Regards, Atis * Using numbers instead of 's' = already tried, no changes * Renaming contexts without underscores = tried it right now, no changes Thanks for all those ideas. Yves. On Thu, 2008-02-07 at 17:39 +0200, Atis Lezdins wrote: On 2/7/08, Yves Räber [EMAIL PROTECTED] wrote: I would have been happy ... but it's not that. This query gives me the right row (I double checked). On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote: On Thursday 07 February 2008 08:05:40 Yves Räber wrote: Hello, I'm having troubles while using the Goto function in a realtime extension. Here is the error message : -- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1) -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context 'script_13_0', but no invalid handler And I definitively have a row in my extensions table with context script_13_0, exten s and priority 1 ! I also tried to goto in another context that is in my extensions.conf file, and it works. Is this a restriction or a bug ? It seems that it's not possible to Goto to another context within the realtime extensions. It's impossible to guess what might be wrong, because you haven't included a dump from your table. Try a: SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0' AND priority='1' If that fails, you have your answer. What version? You could try replacing pipes with commas. Do you have realtime switch statement for script_13_0? Can you try renaming context to not use underscores? Try using not s but any number (and create extension _X.) Regards, Atis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Goto in Realtime extensions
Hello, I'm having troubles while using the Goto function in a realtime extension. Here is the error message : -- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1) -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context 'script_13_0', but no invalid handler And I definitively have a row in my extensions table with context script_13_0, exten s and priority 1 ! I also tried to goto in another context that is in my extensions.conf file, and it works. Is this a restriction or a bug ? It seems that it's not possible to Goto to another context within the realtime extensions. Cheers, Yves. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto in Realtime extensions
I would have been happy ... but it's not that. This query gives me the right row (I double checked). On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote: On Thursday 07 February 2008 08:05:40 Yves Räber wrote: Hello, I'm having troubles while using the Goto function in a realtime extension. Here is the error message : -- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1) -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context 'script_13_0', but no invalid handler And I definitively have a row in my extensions table with context script_13_0, exten s and priority 1 ! I also tried to goto in another context that is in my extensions.conf file, and it works. Is this a restriction or a bug ? It seems that it's not possible to Goto to another context within the realtime extensions. It's impossible to guess what might be wrong, because you haven't included a dump from your table. Try a: SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0' AND priority='1' If that fails, you have your answer. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto in Realtime extensions
* Version: Asterisk 1.4.14 * Commas instead of pipes = already tried, this is not working at all * Realtime switch for script_13_0 = No, should I ? This would be really a shame, I want to use realtime BECAUSE I don't want to play with my extensions.conf file. (I'm building a web interface that has to generate the contexts). * Using numbers instead of 's' = already tried, no changes * Renaming contexts without underscores = tried it right now, no changes Thanks for all those ideas. Yves. On Thu, 2008-02-07 at 17:39 +0200, Atis Lezdins wrote: On 2/7/08, Yves Räber [EMAIL PROTECTED] wrote: I would have been happy ... but it's not that. This query gives me the right row (I double checked). On Thu, 2008-02-07 at 08:36 -0600, Tilghman Lesher wrote: On Thursday 07 February 2008 08:05:40 Yves Räber wrote: Hello, I'm having troubles while using the Goto function in a realtime extension. Here is the error message : -- Executing Goto(SIP/siemens1-081f56b0, script_13_0|s|1) -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context 'script_13_0', but no invalid handler And I definitively have a row in my extensions table with context script_13_0, exten s and priority 1 ! I also tried to goto in another context that is in my extensions.conf file, and it works. Is this a restriction or a bug ? It seems that it's not possible to Goto to another context within the realtime extensions. It's impossible to guess what might be wrong, because you haven't included a dump from your table. Try a: SELECT * FROM extensions_table WHERE exten='s' AND context='script_13_0' AND priority='1' If that fails, you have your answer. What version? You could try replacing pipes with commas. Do you have realtime switch statement for script_13_0? Can you try renaming context to not use underscores? Try using not s but any number (and create extension _X.) Regards, Atis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Goto in Realtime extensions
I'm not using labels at all (but I've also tried with :)) On Thu, 2008-02-07 at 16:39 -0800, Grey Man wrote: Make sure you don't have any labels on the prioritys. When loading extensions from realtime labels aren't supported. Replace: exten = _X.,1(mylabel),... with exten = _X.,1,... You'll have to make your Goto's use the prioritty instead of the label afterward. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to prevent logging of some entries in CDR
Hi On Jan 21, 2008 11:05 PM, Jean-Yves Avenard [EMAIL PROTECTED] wrote: This works great. However in the CDR, than seeing one entry for each call, I see several entries in the CDR Worse, if I do something like: Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]) 40. 2008-01-21 13:59:34 Local/1... 04 MOB. 04 04DialSIP/ipp100SIP/ipp100.1 100 NO ANSWER 00:11 incoming-zap 41. 2008-01-21 13:59:34 SIP/ipp... 04 04 s NO ANSWER 00:11 42. 2008-01-21 13:59:34 SIP/ipp... 04 04 s NO ANSWER 00:11 43. 2008-01-21 13:59:33 Zap/7-1... 04 MOB. 04 04DialLocal/[EMAIL PROTECTED]|10|tr286 ANSWERED00:12 incoming-zap 44. 2008-01-21 13:49:39 Local/1... 04 MOB. 04 04DialSIP/ipp100SIP/ipp100.1 100 NO ANSWER 00:05 102 NO ANSWER 00:05 52. 2008-01-21 13:49:39 Local/1... 04 MOB. 04 04DialSIP/ipp119SIP/ipp119.1 119 NO ANSWER 00:05 No one else is seeing this issue ? Jean-Yves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to prevent logging of some entries in CDR
Hi On Jan 25, 2008 4:58 AM, John Faubion [EMAIL PROTECTED] wrote: I have the same issue but I haven't put much effort into solving it yet. Too many other issues seem to get in the way. If you do, please post your results ! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users