[asterisk-users] SIP Call setup time monitoring in Asterisk
Hi Users, Does Asterisk provide any way to monitor the SIP call setup time between the clients ?? I understand that there is a way to monitor the RTP data flow for jitter and packet losses using *ship show channelstats*. I am looking something on similar lines to monitor call setup time during SIP signalling. Otherwise please suggest other ways of monitoring it ?? Thanks, Abhinav -- _ -- 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] Connect Asterisk to a cell phone
The cellphone can be presented to Asterisk as SIP device using OpenBTS (GSM to SIP conversion). On Tue, Feb 15, 2011 at 10:40 PM, Faisal Hanif fai...@vopium.com wrote: Hi, Your question is not clear but below are possible answers to your question, If you want to attach you cell-phone to asterisk you can simply use chan_mobile. Using Bluetooth with chan_mobile you can connect your Cell-Phone as FXO and your handsfree as FXS port to asterisk. If you are asking about a GSM to SIP gateway then yes there are number of product available that can hold 1-256 SIM and register as SIP gateway to asterisk for incoming and outgoing calls. If you are asking about GSM PCI card then also yes there are PCI cards available for GSM/CDMA/HSPDA for 1-16 SIMs. Can pluged to asterisk PBX machine and used as FXO device. Regards, Faisal Hanif *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *logan *Sent:* Wednesday, February 16, 2011 10:49 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Connect Asterisk to a cell phone Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. -- _ -- 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] Asterisk extension not found problem...
Hi Steve, The asterisk CLI shows the context of caller as below: *moment-portable*CLI sip show user IMSI310410270465840 moment-portable*CLI * Name : IMSI310410270465840 Secret : Not set MD5Secret: Not set Context : sip-external Language : AMA flags: Unknown Transfer mode: open MaxCallBR: 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup: Pickupgroup : Callerid : 2102 ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs Codec Order : (gsm:20) Auto-Framing: No *But when I do dialplan show 2103@sip-external, it returns no dialplan *moment-portable*CLI dialplan show 2103@sip-external There is no existence of 'sip-external' context Command 'dialplan show 2103@sip-external' failed. * I have already created a dialplan in my extensions.conf, I am not sure what is happening here ?? Badly need help in this. Thanks, Abhinav On Tue, Jan 18, 2011 at 8:37 PM, Steve Edwards asterisk@sedwards.comwrote: On Tue, 18 Jan 2011, abhinav anand wrote: The exact error thrown on Asterisk CLI is chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to extension 2103 rejected because extension not found What context does 'sip show user IMSI310410270465840' show? What does 'dialplan show 2103@context-from-previous-command' show? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 extension not found problem...
Hi Steve, Here are the answers to the questions. *1) Do you need to do a 'dialplan reload?'* I don't need to do a dialplan reload. Infact there is no such command as dialplan reload. I simply do a reload each time I make a config change. *2) Are you sure you are editing the extensions.conf that your Asterisk is configured to read?* There are two extensions.conf files present in */etc/asterisk/extensions.conf /home/moment/openbts-uhd/public-trunk/AsteriskConfig/extensions.conf *I am making the changes in /etc/asterisk file. However, when I have tried putting same changes in other file too but again no success. *3) Do you start Asterisk with the ? command line option?* I start Asteisk using sudo asterisk -vvvgcr or sudo asterisk -r. *-c says Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect*. . *4) What is the value of 'astetcdir' in asterisk.conf?* The value is as astetcdir = /etc/asterisk and other values are: [directories](!) ; remove the (!) to enable this astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astdbdir = /var/lib/asterisk astkeydir = /var/lib/asterisk astdatadir = /usr/share/asterisk astagidir = /usr/share/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk Some extra information: - My asterisk version is *Asterisk 1.6.2.5-0ubuntu1.1 built by buildd @ palmer on a i686 running Linux on 2010-07-16 13:24:33 UTC* - I am not able to verify the symlink between the two extensions.conf files Thanks, Abhinav On Wed, Jan 19, 2011 at 2:38 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 19 Jan 2011, abhinav anand wrote: The asterisk CLI shows the context of caller as below: moment-portable*CLI sip show user IMSI310410270465840 Context : sip-external But when I do dialplan show 2103@sip-external, it returns no dialplan moment-portable*CLI dialplan show 2103@sip-external There is no existence of 'sip-external' context Command 'dialplan show 2103@sip-external' failed. I have already created a dialplan in my extensions.conf, I am not sure what is happening here ?? 1) Do you need to do a 'dialplan reload?' 2) Are you sure you are editing the extensions.conf that your Asterisk is configured to read? 3) Do you start Asterisk with the ? command line option? 4) What is the value of 'astetcdir' in asterisk.conf? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 extension not found problem...
'[pbx_ael] Include ='ael-trunkint'[pbx_ael] Ignore pattern = '9' [pbx_ael] moment-portable*CLI * Thanks, Abhinav On Wed, Jan 19, 2011 at 3:44 PM, Steve Edwards asterisk@sedwards.comwrote: Please do not add me or yourself to the address list. We should keep the discussion on the list (and just the list) so it is available to everyone. Also, top-posting is 'frowned upon.' On Wed, 19 Jan 2011, abhinav anand wrote: Here are the answers to the questions. 1) Do you need to do a 'dialplan reload?' I don't need to do a dialplan reload. Infact there is no such command as dialplan reload. I simply do a reload each time I make a config change. What version of Asterisk are you using? 1.2 = 'extensions reload' 1.6 = 'dialplan reload' (I don't have a 1.4 or 1.8 on hand.) If you don't have one of these, something is seriously wrong. 2) Are you sure you are editing the extensions.conf that your Asterisk is configured to read? There are two extensions.conf files present in /etc/asterisk/extensions.conf /home/moment/openbts-uhd/public-trunk/AsteriskConfig/extensions.conf I am making the changes in /etc/asterisk file. However, when I have tried putting same changes in other file too but again no success. 3) Do you start Asterisk with the -C command line option? I start Asteisk using sudo asterisk -vvvgcr or sudo asterisk -r. -c says Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect. The 'r' command line option asks to connect to an existing instance, so this is not the command you use to start Asterisk. The 'upper-case C' command line option allows you to specify location other than /etc/asterisk/ for asterisk.conf. Typing 'echo $(cat /proc/pid-of-asterisk/cmdline)' will show the command line and options Asterisk was started with. 4) What is the value of 'astetcdir' in asterisk.conf? The value is as astetcdir = /etc/asterisk and other values are: [directories](!) ; remove the (!) to enable this astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astdbdir = /var/lib/asterisk astkeydir = /var/lib/asterisk astdatadir = /usr/share/asterisk astagidir = /usr/share/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk Some extra information: - My asterisk version is Asterisk 1.6.2.5-0ubuntu1.1 built by buildd @ palmer on a i686 running Linux on 2010-07-16 13:24:33 UTC So 'dialplan reload' would be the proper command to just reload the dialplan. - I am not able to verify the symlink between the two extensions.conf files If you edit one and your edits don't magically appear in the other, they are not linked. The 'ls' command can also be use to confirm 'linkness.' When you do a 'dialplan show' do you see lines like: 1. mumble-mumble [pbx_config] or 1. mumble-mumble [pbx_ael] or both? (pbx_config means extensions.conf, pbx_ael means extensions.ael) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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 -- ABHINAV ANAND 3GPP Modem Lab., Modem RD Team Telecommunication RD Center Information Communication Business SAMSUNG ELECTRONICS Co., Ltd. SUWON, SOUTH KOREA -- _ -- 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 extension not found problem...
Thanks Steve, I figured out the problem. As you said correctly, *pbx_config.so* was not getting loaded because in my extensions.conf file one extra file extensions.local.conf was included which was actually not present in the directory. I have commented that line and did *module load pbx_config.so* to reload pbx_config.so and now I see both dialplan reload and my sip-external extensions correctly. many many thanks to you for all your pointers and input. I hope this resolves my issue. Thanks to Carlos too for pointing out the reason. Fortunately I am saved from extconfig.conf thing :) Thanks, Abhinav On Wed, Jan 19, 2011 at 4:43 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Wed, 2011-01-19 at 16:40 -0800, Steve Edwards wrote: Un-top-posting... On Wed, 19 Jan 2011, abhinav anand wrote: I am using Asterisk version 1.6.2.5-0. Surprisingly, I don't see dialplan reload. If you do not have 'dialplan reload,' you do not have pbx_config.so loaded. Since pbx_config.so reads extensions.conf, if you don't have it loaded, extensions.conf will not be read. My dialplan show returns some 28 contexts (all pbx_ael and no pbx_config) and looks like this (seems context are read from extensions.ael file only) So, you need to either load pbx_config.so to read your extensions.conf or add the 'sip-external' context to extensions.ael. The last time this happened to me was because extensions.conf had some strange characters that prevented it from loading. Also check that you are not trying to load it from a database by means of extconfig.conf -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 extension not found problem...
Hi All,
I am using Asterisk for one of my projects in OpenBTS. I am having the age
old problem of extension not found when try to make
a call from one registered SIP phone to other registered SIP phone (two
mobile phones connected to Asterisk via OpenBTS).
The exact error thrown on Asterisk CLI is
*chan_sip.c:20039 handle_request_invite: Call from [IMSI310410270465840] to
extension 2103 rejected because extension not found*
I have provisioned for both the phones in *sip.conf* and
*extensions.conf*under context
* [sip-external]* but I suspect whatever entry given in extensions.conf,
that file is not getting parsed and extensions are not read.
I have tried all the methods suggested by others in the Asterisk User
community but still the problem remains same. If anybody knows the solution
to this
one, please let me know.
--
Abhinav
Copied below is my sip.conf and extensions.conf
===
*extensions.conf*
===
[globals]
;Using this Macro
[macro-dialGSM]
exten = s,1,Dial(SIP/${ARG1})
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-CONGESTION,1,Congestion(30)
exten = s-CHANUNAVAIL,1,playback(ss-noservice)
exten = s-CANCEL,1,Hangup
#include extensions.local.conf
[sip-external]
exten = 2101,1,Macro(dialGSM,2101)
exten = 2102,1,Macro(dialGSM,IMSI310410270465840)
exten = 2103,1,Macro(dialGSM,IMSI404864430002302)
; check for local extensions first
include = sip-local
===
*sip.conf*
==
[general]
; Comment these out if no backhaul is available.
; Use the pair with the shortest latency.
;register = kestrel0:v01pt...@sip.ca1.link2voip.com:5060
;register = kestrel0:v01pt...@sip.ca2.link2voip.com:5060
;register = kestrel0:v01pt...@sip.us1.link2voip.com:5060
;register = kestrel0:v01pt...@sip.us2.link2voip.com:5060
;register = kestrel0:v01pt...@sip.nl1.link2voip.com:5060
;register = kestrel0:v01pt...@sip.nl2.link2voip.com:5060
rtpstart=16386
rtpend=16482
relaxdtmf=yes
[softPhone]
callerid=2101
canreinvite=no
type=friend
context=sip-external
allow=ulaw
allow=gsm
host=dynamic
; provisioned Thu Dec 13 17:15:10 2010
[IMSI310410270465840] ; ATnT SIM card IMSI
callerid=2102
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
dtmfmode=info
; provisioned Thu Dec 14 12:15:10 2010
[IMSI404864430002302] ; Vodafone SIM card IMSI
callerid=2103
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
dtmfmode=info
==
--
_
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