Re: [asterisk-users] Asterisk + VOSP account working configuration?
On Tue, 14 Dec 2010 11:19:48 -0600, Lyle Giese l...@lcrcomputer.net
wrote:
You are setting up a SIP trunk from your VOSP provider(whatever VOSP
is). It dials your phone number. So whatever you dial from your cell
phone is the extension that this trunk should land at.
's' is not an extension. It's a placeholder for the steps in your dial plan.
Thanks Lyle, but this document showed a working example using the s
extension, and I did get it working using this:
www.beardy.se/how-to-set-up-a-sip-trunk-in-the-asterisk-pbx
However, I noticed that order is important: When [vosp_incoming] came
before [vosp_outgoing], I got a BUSY signal when calling the VOSP
number, while I could still make outgoing calls to the POTS:
;== sip.conf
[ general]
port = 5060
bindaddr = 0.0.0.0
externip=my public IP
localnet=192.168.0.0/24
disallow=all
allow=ulaw
allow=alaw
allow=gsm
register = myaccount:mypas...@myvosp.com
;IMPORTANT: outgoing must be BEFORE incoming
[vosp_outgoing]
type=peer
host=myvosp.com
username=myaccount
secret=mypasswd
fromuser=myaccount
fromdomain=myvosp.com
nat=yes
canreinvite=no
[vosp_incoming]
type=peer
host=myvosp.com
nat=yes
canreinvite=no
context=from_vosp
[6011]
type=friend
secret=my_password
host=dynamic
context=internal
;calls go through Asterisk on same LAN
nat=no
;== extensions.conf
[general]
static=yes
writeprotect=yes
clearglobalvars=no
autofallthrough=yes
[from_vosp]
exten = s,1,Dial(SIP/6011)
exten = s,n,Hangup()
[to_vosp]
;All numbers starting with 0 are sent to VOSP
exten = _0.,1,Dial(SIP/ippi_outgoing/${EXTEN})
exten = _0.,n,Hangup()
[internal]
exten = 6011,1,Dial(SIP/6011)
exten = 6011,n,Hangup()
exten = 6012,1,Dial(SIP/6012)
exten = 6012,n,Hangup()
include = to_vosp
;==
If someone knows of a thorough article/book on how configuration files
work in Asterisk, I'm interested.
Thank you.
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[asterisk-users] Asterisk Community Mailing Lists Service Disruption
The system that hosts the Asterisk community mailing lists (lists.digium.com) experienced some failures yesterday, and as a result the lists have been moved to a new system (with the same name). During this process, it is possible that outbound messages queued for delivery to some list subscribers were lost, and it is also possible that some messages sent by subscribers were never received by the list server. If you feel you may have missed some mailing list messages, feel free to review the list archives as they should contain all messages that were actually received by the list server. We apologize for any inconvenience you may experience due to this disruption. -- _ -- 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] Transferring problem within Queues
Hi We are using asterisk 1.4.17 for the apt repository on an Ubuntu server and we're getting an odd problem with one customer using a Queue The queue is called in the dialplan with the options Tn The queue only has one member. Occasionally and starting to get more frequently the caller ends up being initially answered by the wrong extension (i.e. one that is not a member of the queue) Has anyone ever heard of an issue where on call in a queue is transferred but the transfer is actually actioned on a different call that is waiting in the same queue? Thanks in Advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Which version to use: 1.4 or 1.6 or 1.8
Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? 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
[asterisk-users] Asterisk 1.8 with web-meetme crash
Hi All, Anyone out there successfully tested Asterisk 1.8 with Web-Meetme 4.0v in my case my asterisk got crashed when i dialing conf room number. Best, S -- _ -- 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 error - 1.6.2 SVN - voicemail files corrupted
Hi Tilghman, I am indeed still seeing this issue (emails missing in sequence, and therefore voicemail box not readable), and I have absolutely no third-party vendor solution playing with voicemails. How do I find whether this was a simple bug that was found and fixed in between official versions? (since I am using SVN?) Or how do I debug and find what was the root cause of the issue? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, December 03, 2010 9:49 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted Hi Tilghman, This particular customer was one of my less sophisticated customer, and I know for sure he isn`t using anything else than Voicemailmain. Not even the basic voicemail to email function. But I will keep an eye opened for any future problem. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, December 03, 2010 4:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk error - 1.6.2 SVN - voicemail files corrupted On Thursday 02 December 2010 18:56:44 Mike wrote: 1) How do I fix this? I don't mind manually fixing it when it happens, but what's wrong exactly? There should not be anything within the Asterisk process to cause this. However, I _have_ seen this exact issue with certain 3rd party vendors that supply a tool for checking voicemail via a web interface. The offending tools make no effort to reorder the messages after certain messages are deleted, which is a really bad thing to do. If this is, in fact, the issue, please ask the vendor to fix the interface, because in the current form, it is severely broken behavior. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- 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
[asterisk-users] res_odbc dependeny issue
Hi All, I have issue with res_odbc.so module Asterisk 1.8 not allowing me to enable because its depended on generic_odbc and ltdl I did install unixodbc and ltdl but still same error Thanks, Satish -- _ -- 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] Recommendation for a Linux based SCADA
Hi list, For a telecom project I need to setup a SCADA solution. I don't have any previous experience in this type of monitoring and automization. I'll be using SNMP data from asterisk servers and endpoints. If anybody has any suggestion which SCADA software can fit in such a VoIP solution, your guidance will be highly appreciated. Thanks, Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.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] Asterisk 1.8.1.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.1.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.1.1 resolves two issues reported by the community since the release of Asterisk 1.8.1. * Don't crash after Set(CDR(userfield)=...) in ast_bridge_call. Instead of setting peer-cdr = NULL, set it to not post. (Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) * Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue. (Closes issue #18412. Reported by nevermind_quack. Patched by dvossel) For a full list of changes in this release candidate, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1.1 Thank you for your continued support of Asterisk! -- _ -- 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] Echo Cancellation Problem - Invalid Argument?!?
Greetings folks- I'm experiencing issues with a freshly installed box. When a call comes in via PRI (Sangoma AFT-A104), I see this in my logs: [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 12 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 8 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 10 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 9 (Invalid argument) Relevant components: Asterisk: Asterisk SVN-trunk-r290509 built by root @ prigw01 on a i686 running Linux on 2010-11-30 22:12:05 UTC DAHDI: dahdi-linux-complete-2.4.0+2.4.0 LibPRI: libpri-1.4.11.5 Wanpipe: wanpipe-3.5.18 Kernel: Linux prigw01 2.6.32-24-generic #39-Ubuntu SMP Wed Jul 28 06:07:29 UTC 2010 i686 GNU/Linux The card does not have a hardware echo canceler. It should use MG2 as specified in DAHDI's system.conf: #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit #autogenrated on 2010-12-08 #Dahdi Channels Configurations #For detailed Dahdi options, view /etc/dahdi/system.conf.bak loadzone=us defaultzone=us #Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 span=1,1,0,esf,b8zs bchan=1-23 #dchan=24 echocanceller=mg2,1-23 hardhdlc=24 And, from chan_dahdi.conf: ;Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 switchtype=national context=ldrouted group=1 echocancel=yes signalling=pri_net channel =1-23 Any thoughts, pointers, suggestions? The echo is horrible, please help me make it stop. :-) --Tim -- _ -- 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] res_odbc dependeny issue
On Wednesday 15 December 2010 13:53:12 satish patel wrote: I have issue with res_odbc.so module Asterisk 1.8 not allowing me to enable because its depended on generic_odbc and ltdl I did install unixodbc and ltdl but still same error Make sure you re-run ./configure after you add/remove packages, as the configure script is what determines what packages are found. Additionally, ensure you installed the -devel (-dev on Debian/Ubuntu) packages, as it is the headers in these packages which are needed. -- Tilghman -- _ -- 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 + VOSP account working configuration?
Le 15/12/2010 15:21, Gilles a écrit : [...] ;IMPORTANT: outgoing must be BEFORE incoming [vosp_outgoing] type=peer host=myvosp.com username=myaccount secret=mypasswd fromuser=myaccount fromdomain=myvosp.com nat=yes canreinvite=no [vosp_incoming] type=peer host=myvosp.com nat=yes canreinvite=no context=from_vosp Why 2 context? Todays Asterisk versions only needs one peer context for incoming/outgoing. Something like [vosp] type=peer host=myvosp.com username=myaccount secret=mypasswd fromuser=myaccount fromdomain=myvosp.com nat=yes canreinvite=no context=from_vosp -- Daniel -- _ -- 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] Which version to use: 1.4 or 1.6 or 1.8
What are your needs? Perhaps use stable 1.4 if the provided features suffice you. Am 15.12.2010 15:46, schrieb bilal ghayyad: Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? 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 -- Thorsten Göllner OVM Office Voice Media GmbH Herderstrasse 68 40237 Düsseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54 -- _ -- 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] Which version to use: 1.4 or 1.6 or 1.8
there is no reason not to use 1.8 when you start a new installation. 1.8 is the new five years long term support version Kristijan 2010/12/15 bilal ghayyad bilmar...@yahoo.com: Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? 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 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] Junghanns OctoBRI - Invalid sync priority
Hi, Looking at dmesg, I can see : [ 45.213205] wcb4xxp :01:07.0: Span 5 has invalid sync priority (5), removing from sync source list [ 45.213205] wcb4xxp :01:07.0: Span 6 has invalid sync priority (6), removing from sync source list [ 45.213205] wcb4xxp :01:07.0: Span 7 has invalid sync priority (7), removing from sync source list [ 45.213205] wcb4xxp :01:07.0: Span 8 has invalid sync priority (8), removing from sync source list My setup is : Lenny asterisk 1.6.1.18 dahdi svn rev 9503 libpri 1.4.10.2 #lspci -v -n ... 01:07.0 0204: 1397:16b8 (rev 01) Subsystem: 1397:b552 Flags: medium devsel, IRQ 17 I/O ports at 3400 [size=8] Memory at d011 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Kernel driver in use: wcb4xxp Kernel modules: wcb4xxp Though this doesn't seem cause much trouble, is this a bug ? 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] Call sip:u...@domain.com?
Hello At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT set up with a VOSP trunk that I can use to make/receive calls to/from the PSTN. Now, I'd like to be able to call any number on the Net that is advertised as sip:u...@domain.com, such as those: www.voip-info.org/wiki/view/Phone+Numbers Do I need to register a second trunk (FWD, etc.) through which those calls will be made? Can't my VOSP perform both tasks (landline + Internet calls)? Can I just let my Asterisk server connect to the remote SIP server through the SRV DNS record and have it dial the extension? Any example appreciated, thank you. -- _ -- 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] setting up callerid
Tried the following but no luck:
exten = _53.,1,Set(CALLERID(num)=473520)
exten = _53.,n,Dial(SIP/${ext...@ss74)
I am still passing IMSI310410381554227 as the CALLERID.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
Thanks,
Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten
Göllner
Sent: Monday, December 13, 2010 4:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] setting up callerid
Am 12.12.2010 20:49, schrieb dave george:
I am using Asterisk 1.6.2.5-0 running on ubuntu and I have a problem
passing called ID on calls to the PSTN
When I make a call to the PSTN the caller-Id is showing up as
IMSI310410381554227
I want the number set in the callerid field to show up.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
I use the following in extensions.conf to dial:
exten = _45.,1,Dial(SIP/${ext...@ss72)
Thanks,
Dave
Take a look:
http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerID
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[asterisk-users] chan_iax2.c handle_call_token: Call rejected, CallToken Support required
I had two asterisk servers connected with each other, both were 1.4.22 but I've upgraded one to 1.4.37 and now I get a message when I try call from asterisk-1.4.22 to asterisk-1.4.37 ERROR[4539]: chan_iax2.c:4330 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.140.1 in the calltokenoptional list or setting user guest requirecalltoken=no on asterisk-1.4.37 I have in iax.conf (requirecalltoken=no) [clinic_server] type=friend host=dynamic context=internal disallow=all allow=ulaw allow=alaw allow=g729 requirecalltoken=no and I can call to asterisk-1.4.22 but asterisk-1.4.22 can not connect to 1.4.37 (as error listed above) Are those two asterisks compatible? I've tried placing requirecalltoken=no in iax.conf on asterisk-1.4.22 but it is not working. Any suggestion. -- Joseph -- _ -- 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] Which version to use: 1.4 or 1.6 or 1.8
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad Sent: Wednesday, December 15, 2010 8:47 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8 Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? Regards Bilal 1.4 is the defacto stable release at present (although some folks still use 1.2 or even 1.0). 1.4 and 1.6 both will reach end-of-life in April 2012 (1.6.0 has already reached EOL as well as 1.2). 1.8 is scheduled for EOL in October 2015. Personal experience and my readings here have convinced me to jump straight from 1.4 to 1.8 for my new installations. If you need a bell or whistle that 1.4 does not offer, I'd recommend 1.8. -- _ -- 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 1.8 with web-meetme crash
I'm currently using Web-Meetme 4.0.2 with Asterisk 1.8.0 with no problems. Maybe you could post more information about the crash? On Wed, Dec 15, 2010 at 11:46 AM, satish patel satish...@hotmail.com wrote: Hi All, Anyone out there successfully tested Asterisk 1.8 with Web-Meetme 4.0v in my case my asterisk got crashed when i dialing conf room number. Best, S -- _ -- 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] Transferring problem within Queues [edit] Transferring problem with BLF buttons
On Wed, 2010-12-15 at 14:33 +, Ishfaq Malik wrote: Hi We are using asterisk 1.4.17 for the apt repository on an Ubuntu server and we're getting an odd problem with one customer using a Queue The queue is called in the dialplan with the options Tn The queue only has one member. Occasionally and starting to get more frequently the caller ends up being initially answered by the wrong extension (i.e. one that is not a member of the queue) Has anyone ever heard of an issue where on call in a queue is transferred but the transfer is actually actioned on a different call that is waiting in the same queue? Thanks in Advance Ish I've been doing my own testing so can add some more information to my initial problem. I have set up my phone to use hints/subscriptions for the purpose of BLF Call A comes in and is answered by User 1 Call B comes in and is left ringing User 1 uses a BLF programmed button on a snom phone to call user 2 User 2 answers and accepts the call User 1 presses transfer on their phone Call B (which still has not been answered) is transferred to User 2 and not call A which stays on hold! If I do the same as above but press hold and dial the extension for user 2 and then press transfer call A is transferred to user 2 which is what I think we all would consider to be the correct behaviour. Has anyone ever come across this and found a reason (and possibly solution) for this issue? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Two asterisk servers, two different service providers
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eddie Mikell
Sent: Wednesday, December 15, 2010 7:39 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Two asterisk servers,two different service
providers
All:
I am looking to install another asterisk server in an office located in
a different part of the country.
I think I can configure the sip and extension conf files, so that the
internal phones at the two locations can call each other.
My question is this, how do I properly configure the sip file for a
different provider at the new location? Can I use a different register
statement for the provider at the new location?
Can someone point me to some sample conf files that do this?
Thanks for all help, AND non smart aleck RTFM answers.
Eddie
For question 1 - the provider should give you the credentials you need for
sip.conf (mine did).
For question 2 - you can have as many register statements as you want.
These are two different Asterisk installs and two different SIP
lines/trunks.
What you're doing is reasonably simple. You have office A that uses a SIP
trunk and SIP lines and office B that uses a different SIP trunk and SIP
lines. How I would do it is like this:
Office A uses extensions 1000-1999
Office B uses extensions 2000-2999
The inherent problem you would have is that you can't pass information in
the dial string. The work-around would be to use a small IVR in each office
to do an automated redial. So if I'm in office A and I want to call
somebody from Office B, I dial 2001, get a response to reenter the number
and dial 2001 again and am connected. The cleaner solution would be to
have an IAX connection between the two offices.
Extensions.conf for office A
Exten = 1XXX,1,Dial(SIP/${EXTEN},20,MKkTt)
Exten = _1.,1,Dial(SIP/+${ext...@provider,30,MKkTt)
Exten = 2XXX,1,Dial(SIP/5551...@provider,30,MKkTt) 5551212 is office B's
number
Exten = s/5551313,1,Goto(callfromb,s,1)
[callfromb]
Exten = s,1,read(xext,enter_extension,4,skip,1,10)
Exten = s,n,Dial(SIP,${xext},20,MKkTt)
Extensions.conf for office B
Exten = 2XXX,1,Dial(SIP/${EXTEN},20,MKkTt)
Exten = _1.,1,Dial(SIP/+${ext...@provider,30,MKkTt)
Exten = 1XXX,1,Dial(SIP/5551...@provider,30,MKkTt) 5551313 is office A's
number
Exten = s/5551212,1,Goto(callfroma,s,1)
[callfroma]
Exten = s,1,read(xext,enter_extension,4,skip,1,10)
Exten = s,n,Dial(SIP,${xext},20,MKkTt)
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Re: [asterisk-users] Asterisk 1.6.2.10 video
1. Per http://www.voip-info.org/wiki/view/Asterisk+video: Asterisk does not provide any video transcoding capabilities 2. You can turn off video support on a peer like this: disallow=h261 disallow=h263 disallow=h263p From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Monday, December 13, 2010 3:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.2.10 video Hello, 1. is it possible that Asterisk does not translate between codecs H263 and H264 ? 2 If I set videosupport=yes in sip.conf [general], can I turn off the video support on a peer ? Kind regards, Jonas. -- _ -- 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] Dialplan not found
Hi there! Anybody knows why I am receiving this output from CLI: No such command 'dialplan reload' (type 'core show help dialplan reload' for other possible commands) Look like asterisk dont see dialplan? Is it possible to restart it ? Thansk Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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] app_voicemail: 3 for advanced options does not have an effect _while_ the vm-message is played
3 for advanced options does not have an effect _while_ the vm-message is played. all other options like 7 delete or 6 next message are working. after the vm-message is played, it's working. the question: is this intentional? or a bug? the available documentation does not describe this case. Kristijan -- _ -- 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] Which version to use: 1.4 or 1.6 or 1.8
On 10-12-15 09:46 AM, bilal ghayyad wrote: Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? It depends on your required usage (features available in version) and your required support level. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for information on the support level and time for each of the branches. Leif. -- _ -- 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] res_odbc dependeny issue
On Wed, Dec 15, 2010 at 1:53 PM, satish patel satish...@hotmail.com wrote: Hi All, I have issue with res_odbc.so module Asterisk 1.8 not allowing me to enable because its depended on generic_odbc and ltdl I did install unixodbc and ltdl but still same error Thanks, Satish Try installing libtool-ltdl-devel unixODBC-devel (or whatever the development packages are called on your distro, these are examples from CentOS 5.4). -- Thanks, --Warren Selby, dCAP http://www.selbytech.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] Two asterisk servers, two different service providers
Here is how I would do it: First, come up with a numbering scheme. For instance, all extensions in location 1 are 1xxx, extensions in location 2 are 2xxx. External calls are 9xxx-xxx- In Location 1: Sip Trunk 1 - goes to Location 2. The dial plan uses it as outgoing trunk for all calls with 2xxx Sip Trunk 2 - goes to Provider 1. The dial plan uses it as outgoing trunk for all calls starting with 9 In Location 2, you do basically the same thing in reverse. Sip Trunk 1 - goes to Location 1. The dial plan uses it as outgoing trunk for all calls with 1xxx Sip Trunk 2 - goes to Provider 2. You can probably also use both providers as backup -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eddie Mikell Sent: Wednesday, December 15, 2010 5:39 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Two asterisk servers, two different service providers All: I am looking to install another asterisk server in an office located in a different part of the country. I think I can configure the sip and extension conf files, so that the internal phones at the two locations can call each other. My question is this, how do I properly configure the sip file for a different provider at the new location? Can I use a different register statement for the provider at the new location? Can someone point me to some sample conf files that do this? Thanks for all help, AND non smart aleck RTFM answers. Eddie -- _ -- 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] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI
Hi all, Last night I went to replace an Asterisk 1.4 + mISDN + b410p box with Asterisk 1.6.2 and DAHDI BRI - to no avail. I had two servers so copied network setting etc from the working one, moved the card across, ran dahdi_genconf etc and it didn't work. Here's the console output with notices disabled: lucas*CLI pri intense debug span 1 Enabled debugging on span 1 [Dec 16 18:59:46] ERROR[12785]: chan_dahdi.c:12630 dahdi_pri_error: Unable to receive TEI from network in state 2(Assign awaiting TEI)! Changing from state 2(Assign awaiting TEI) to 1(TEI unassigned) [Dec 16 18:59:46] ERROR[12787]: chan_dahdi.c:12630 dahdi_pri_error: Unable to receive TEI from network in state 2(Assign awaiting TEI)! [Dec 16 18:59:46] ERROR[12788]: chan_dahdi.c:12630 dahdi_pri_error: Unable to receive TEI from network in state 2(Assign awaiting TEI)! [Dec 16 18:59:46] ERROR[12786]: chan_dahdi.c:12630 dahdi_pri_error: Unable to receive TEI from network in state 2(Assign awaiting TEI)! TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 04 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 4(TEI Identity Check Request) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 06 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 6(TEI Identity Remove) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 06 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 6(TEI Identity Remove) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 04 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 4(TEI Identity Check Request) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 06 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 6(TEI Identity Remove) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 06 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 6(TEI Identity Remove) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 04 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 4(TEI Identity Check Request) Ri: 0 Ai: 127 E:1 Received MDL message = Here's the console output with notices enabled: [Dec 16 19:02:34] NOTICE[12786]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Dec 16 19:02:34] NOTICE[12785]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Dec 16 19:02:34] NOTICE[12786]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Dec 16 19:02:34] NOTICE[12785]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Dec 16 19:02:34] NOTICE[12786]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Dec 16 19:02:34] NOTICE[12785]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Dec 16 19:02:34] NOTICE[12786]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Dec 16 19:02:35] NOTICE[12786]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Dec 16 19:02:35] NOTICE[12785]: chan_dahdi.c:12931
[asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI
Hi all, (Sorry if this is the second email - didn't see the first) Last night I went to replace an Asterisk 1.4 + mISDN + b410p box with Asterisk 1.6.2 and DAHDI BRI - to no avail. I had two servers so copied network setting etc from the working one, moved the card across, ran dahdi_genconf etc and it didn't work. Here's the console output with notices disabled: lucas*CLI pri intense debug span 1 Enabled debugging on span 1 [Dec 16 18:59:46] ERROR[12785]: chan_dahdi.c:12630 dahdi_pri_error: Unable to receive TEI from network in state 2(Assign awaiting TEI)! Changing from state 2(Assign awaiting TEI) to 1(TEI unassigned) [Dec 16 18:59:46] ERROR[12787]: chan_dahdi.c:12630 dahdi_pri_error: Unable to receive TEI from network in state 2(Assign awaiting TEI)! [Dec 16 18:59:46] ERROR[12788]: chan_dahdi.c:12630 dahdi_pri_error: Unable to receive TEI from network in state 2(Assign awaiting TEI)! [Dec 16 18:59:46] ERROR[12786]: chan_dahdi.c:12630 dahdi_pri_error: Unable to receive TEI from network in state 2(Assign awaiting TEI)! TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 04 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 4(TEI Identity Check Request) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 06 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 6(TEI Identity Remove) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 06 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 6(TEI Identity Remove) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 04 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 4(TEI Identity Check Request) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 06 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 6(TEI Identity Remove) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 06 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 6(TEI Identity Remove) Ri: 0 Ai: 127 E:1 Received MDL message TEI: 0 State 1(TEI unassigned) V(A)=0, V(S)=0, V(R)=0 K=1, RC=0, l3_initiated=0, reject_except=0, ack_pend=0 T200_id=0, N200=3, T203_id=0 [ fe ff 03 0f 00 00 04 ff ] Unnumbered frame: SAPI: 63 C/R: 1 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data MDL Message: 4(TEI Identity Check Request) Ri: 0 Ai: 127 E:1 Received MDL message = Here's the console output with notices enabled: [Dec 16 19:02:34] NOTICE[12786]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Dec 16 19:02:34] NOTICE[12785]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Dec 16 19:02:34] NOTICE[12786]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Dec 16 19:02:34] NOTICE[12785]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Dec 16 19:02:34] NOTICE[12786]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Dec 16 19:02:34] NOTICE[12785]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Dec 16 19:02:34] NOTICE[12786]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2 [Dec 16 19:02:35] NOTICE[12786]: chan_dahdi.c:12931 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2
Re: [asterisk-users] Call sip:u...@domain.com?
Just add something like this to your dialplan: exten=1234,1,Dial(SIP/u...@domain.com) Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Thursday, December 16, 2010 5:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call sip:u...@domain.com? Hello At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT set up with a VOSP trunk that I can use to make/receive calls to/from the PSTN. Now, I'd like to be able to call any number on the Net that is advertised as sip:u...@domain.com, such as those: www.voip-info.org/wiki/view/Phone+Numbers Do I need to register a second trunk (FWD, etc.) through which those calls will be made? Can't my VOSP perform both tasks (landline + Internet calls)? Can I just let my Asterisk server connect to the remote SIP server through the SRV DNS record and have it dial the extension? Any example appreciated, thank you. -- _ -- 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 + VOSP account working configuration?
On Thu, 16 Dec 2010 10:06:35 +0100, Administrator TOOTAI ad...@tootai.net wrote: Why 2 context? Todays Asterisk versions only needs one peer context for incoming/outgoing. Something like I tried combining the two sections in sip.conf, but get a BUSY signal for incoming calls from the PSTN. Could it be because I'm running Asterisk 1.4? www.ippi.fr/index.php?page=sip_parameter (scroll down to PBX Asterisk - TrixBox) Also, it's important to have the Outgoing part come before the Incoming. Otherwise, I also get a BUSY signal for incoming calls. BTW, why do we need to define the login/password twice, once with register and a second time in a [_outgoing] section? Thank you. -- _ -- 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] setting up callerid
Dave,
Can you capture the cli output and the sip debug of the call not doing what
it's supposed to?
Thanks,
--Warren Selby, dCAP
On Dec 16, 2010, at 6:52 AM, dave george dgeo...@teletoneinc.com wrote:
Tried the following but no luck:
exten = _53.,1,Set(CALLERID(num)=473520)
exten = _53.,n,Dial(SIP/${ext...@ss74)
I am still passing IMSI310410381554227 as the CALLERID.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
Thanks,
Dave
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thorsten Göllner
Sent: Monday, December 13, 2010 4:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] setting up callerid
Am 12.12.2010 20:49, schrieb dave george:
I am using Asterisk 1.6.2.5-0 running on ubuntu and I have a problem passing
called ID on calls to the PSTN
When I make a call to the PSTN the caller-Id is showing up as
IMSI310410381554227
I want the number set in the callerid field to show up.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
disallow=all
allow=gsm
host=dynamic
dtmfmode=info
I use the following in extensions.conf to dial:
exten = _45.,1,Dial(SIP/${ext...@ss72)
Thanks,
Dave
Take a look:
http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerID
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Re: [asterisk-users] Which version to use: 1.4 or 1.6 or 1.8
linuxinnovations.com is also a good place to seek out the differences between the versions. On Thu, Dec 16, 2010 at 7:00 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 10-12-15 09:46 AM, bilal ghayyad wrote: Hi All; I need to know which version of asterisk to use, if to be 1.4 or 1.6 or 1.8? For example, when to decide that I have to go for 1.6 or I have to go for 1.8? It depends on your required usage (features available in version) and your required support level. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for information on the support level and time for each of the branches. Leif. -- _ -- 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] How to find , internal, external inbound or outbound
Hi Does anyone knows how to find out a call in a asterisk is external incoming ,external out going or internal Thanks Nikhil -- _ -- 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] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI
Hi, Did you use libpri 1.4.11.5 or 1.4.12-beta ? Recently l tried 1.4.11.5 on a live system and it failed (Asterisk 1.6.1.18 and dahdi trunk, Junghanns QuadBRI, PtmP lines). Going back to 1.4.11.2 solved it. Unfortunately, I couldn't note what error message were then generated. Hope this helps. Cheers -- _ -- 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
