[asterisk-users] sip realtime question
Hi everybody,
First of all i am sorry my English :)
i want to configure my asterisk server as a sip server that stores sip users in
the mysql database connecting directly over odbc driver. My odbc configuration
works as below
[r...@ao042 asterisk]# isql -v asterisk
+---+
| Connected!|
| |
| sql-statement |
| help [tablename] |
| quit |
| |
+---+
and i did refer to the site
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip . my res_odbc file as;
[r...@ao042 asterisk]# cat res_odbc.conf
[asterisk]
enabled = yes
dsn = asterisk
username = asterisk
password = ***
pre-connect = yes
[r...@ao042 asterisk]# grep -Ev '^;|^$' extconfig.conf
[settings]
sipusers = odbc,asterisk,sip_buddies
sippeers = odbc,asterisk,sip_buddies
and i created the asterisk database with sip_buddies table.
Here is my problem:
In asterisk console when i run the following command i get the answer,
ao042*CLI realtime load sipusers name 100
Column Name Column Value
id 1
name 100
host dynamic
nat no
type friend
cancallforward yes
canreinvite yes
secret Deneme01
disallow all
allow g729
allow ilbc
allow gsm
allow ulaw
allow alaw
port 5060
regseconds 0
lastms 0
username 100
but the following commands returns nothing
ao042*CLI sip show users
Username Secret Accountcode Def.Context
ACL NAT
ao042*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
besides it does not query anything. So what am i missing? Is there anything
that i should mentioned in the sip.conf file?
by the way i am using RHEL 5.4 with 2.6.18-164.el5 kernel -
asterisk16-1.6.0.17-1_centos5 rpm
Any help would be appreciated...
--
Emre Kurnaz
ITU/BIDB | Istanbul Technical University /
Information Technologies Office
Sistem Destek Grubu| System Support Team
RHCE : 805009174841679
Yarı Zamanlı Öğrenci Koordinatörü | Part-Time Student Manager
kurn...@itu.edu.tr
+90 0212 285 3930
___
-- 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 realtime question
I am not sure, but I think you will get nothing with those commands if realtime
cathing is not set.
--Original Message--
From: Emre Kurnaz
Sender: asterisk-users-boun...@lists.digium.com
To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] sip realtime question
Sent: Dec 11, 2009 4:02 AM
Hi everybody,
First of all i am sorry my English :)
i want to configure my asterisk server as a sip server that stores sip users in
the mysql database connecting directly over odbc driver. My odbc configuration
works as below
[r...@ao042 asterisk]# isql -v asterisk
+---+
| Connected!|
| |
| sql-statement |
| help [tablename] |
| quit |
| |
+---+
and i did refer to the site
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip . my res_odbc file as;
[r...@ao042 asterisk]# cat res_odbc.conf
[asterisk]
enabled = yes
dsn = asterisk
username = asterisk
password = ***
pre-connect = yes
[r...@ao042 asterisk]# grep -Ev '^;|^$' extconfig.conf
[settings]
sipusers = odbc,asterisk,sip_buddies
sippeers = odbc,asterisk,sip_buddies
and i created the asterisk database with sip_buddies table.
Here is my problem:
In asterisk console when i run the following command i get the answer,
ao042*CLI realtime load sipusers name 100
Column Name Column Value
id 1
name 100
host dynamic
nat no
type friend
cancallforward yes
canreinvite yes
secret Deneme01
disallow all
allow g729
allow ilbc
allow gsm
allow ulaw
allow alaw
port 5060
regseconds 0
lastms 0
username 100
but the following commands returns nothing
ao042*CLI sip show users
Username Secret Accountcode Def.Context
ACL NAT
ao042*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
besides it does not query anything. So what am i missing? Is there anything
that i should mentioned in the sip.conf file?
by the way i am using RHEL 5.4 with 2.6.18-164.el5 kernel -
asterisk16-1.6.0.17-1_centos5 rpm
Any help would be appreciated...
--
Emre Kurnaz
ITU/BIDB | Istanbul Technical University /
Information Technologies Office
Sistem Destek Grubu| System Support Team
RHCE : 805009174841679
Yar¹ Zamanl¹ Ö»renci Koordinatörü | Part-Time Student Manager
kurn...@itu.edu.tr
+90 0212 285 3930
___
-- 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
Saludos,
Juan E. Rodríguez
___
-- 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] How to get LEG B channel info?
Hello,
How can I go to the Leg B channel in Asterisk Dialplan _after_ call ends?
I can use Dial G option to go to Leb B channel when call is answered, but
how to go here when call ends?
Is here any option/function in Dial Plan?
Or should I use ast_bridged_channel(chan) to get bridged channel and try to
retrieve data I need from internal structures using custom c module and
Asterisk API?
I'm trying to retrieve ${CHANNEL(rtpqos,audio,all)} for Leg B.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip realtime question
Emre Kurnaz wrote:
Hi everybody,
First of all i am sorry my English :)
i want to configure my asterisk server as a sip server that stores sip users
in the mysql database connecting directly over odbc driver. My odbc
configuration works as below
[r...@ao042 asterisk]# isql -v asterisk
+---+
| Connected!|
| |
| sql-statement |
| help [tablename] |
| quit |
| |
+---+
and i did refer to the site
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip . my res_odbc file
as;
[r...@ao042 asterisk]# cat res_odbc.conf
[asterisk]
enabled = yes
dsn = asterisk
username = asterisk
password = ***
pre-connect = yes
[r...@ao042 asterisk]# grep -Ev '^;|^$' extconfig.conf
[settings]
sipusers = odbc,asterisk,sip_buddies
sippeers = odbc,asterisk,sip_buddies
and i created the asterisk database with sip_buddies table.
Here is my problem:
In asterisk console when i run the following command i get the answer,
ao042*CLI realtime load sipusers name 100
Column Name Column Value
id 1
name 100
host dynamic
nat no
type friend
cancallforward yes
canreinvite yes
secret Deneme01
disallow all
allow g729
allow ilbc
allow gsm
allow ulaw
allow alaw
port 5060
regseconds 0
lastms 0
username 100
but the following commands returns nothing
ao042*CLI sip show users
Username Secret Accountcode Def.Context
ACL NAT
ao042*CLI sip show peers
Name/username HostDyn Nat ACL Port Status
Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
besides it does not query anything. So what am i missing? Is there anything
that i should mentioned in the sip.conf file?
by the way i am using RHEL 5.4 with 2.6.18-164.el5 kernel -
asterisk16-1.6.0.17-1_centos5 rpm
Any help would be appreciated...
in sip.conf you need to change the following
rtcachefriends=yes
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
___
-- 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] ANNOUNCE: New version of Activa TAPI driver
hello, there is new version of the best open source TAPI driver for Asterisk - Activa 1.6.1 * NEW: Asterisk 1.6 compatibility (partially sponsored by IPEX a.s. http://www.ipex.cz) * NEW: FEATURE_CODES standardization for AgentACD integration login, logout, ready, notReady. * NEW: ActivaTSP x64 version. * New: Windows 2008 Server compatibility. * CHANGE: Some performance optimization. * FIX: SIP/ Dns can generate void extensions. * FIX: in process dn expresion, the duplicate filter deletes non duplicate entries. download: http://sourceforge.net/projects/activa/files/ doc: http://activa.sourceforge.net/readme.html many thanks to Activa Team --- Marek Cervenka === ___ -- 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] ANNOUNCE: New version of Activa TAPI driver
2009/12/11 marek cervenka cerv...@fpf.slu.cz hello, there is new version of the best open source TAPI driver for Asterisk - Activa 1.6.1 * NEW: Asterisk 1.6 compatibility (partially sponsored by IPEX a.s. http://www.ipex.cz) * NEW: FEATURE_CODES standardization for AgentACD integration login, logout, ready, notReady. * NEW: ActivaTSP x64 version. * New: Windows 2008 Server compatibility. * CHANGE: Some performance optimization. * FIX: SIP/ Dns can generate void extensions. * FIX: in process dn expresion, the duplicate filter deletes non duplicate entries. download: http://sourceforge.net/projects/activa/files/ doc: http://activa.sourceforge.net/readme.html many thanks to Activa Team --- Marek Cervenka === Congratulations for this release !! I'm not sure this list is the right place to ask but is it possible, to install ActivaTSP on a single Windows server and simply configure each Windows XP/Vista station's TAPI stack to use this server resource (avoiding adding any software on client PCs) ? ___ -- 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] Calls Dropping
Hello, We have a problem that calls seem to be dropping for no reason. Is there any way to write a debug log to disk so that I can check it as soon as a call is lost? It happens randomly once or twice a day to different callers. Many thanks Dan ___ -- 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] ANNOUNCE: New version of Activa TAPI driver
Hi all, As member of Activa Team I invite all interested in an Asterisk Tapi Service Provider to integrate with TAPI applications to try activaTSP and send us feedback in our forum / bugtracker. Oliver, Activa TSP was developed by ICR and given to the Community to facilitate the creation of telephony applications. ICR is the first company to openly release a professional free software solution, EVOLUTION Call Center. This solution have a server piece that use an unique activaTSP to control the entire callcenter and then the agent applications are connected to the server... If anyone is interested, EVOLUTION Call Center is available in Spanish (www.evolutioncallcenter.com). I have information that English version is planned to be released in Q1 2010. I can tell you that ActivaTSP is a powerfull TSP that is implemented to be used as a simple client that control one extension or as a server piece that control an entire callcenter by an unique server that have connected clients. An alternative to don't install the TSP in all client PC you can try to configure a Remote Service Provider, but I never tested it. I think the best place to discuss the usage of the TSP is in the activaTSP forum (http://sourceforge.net/projects/activa/forums). Josep De: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] En nombre de Olivier Enviado el: viernes, 11 de diciembre de 2009 11:54 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] ANNOUNCE: New version of Activa TAPI driver 2009/12/11 marek cervenka cerv...@fpf.slu.cz hello, there is new version of the best open source TAPI driver for Asterisk - Activa 1.6.1 * NEW: Asterisk 1.6 compatibility (partially sponsored by IPEX a.s. http://www.ipex.cz) * NEW: FEATURE_CODES standardization for AgentACD integration login, logout, ready, notReady. * NEW: ActivaTSP x64 version. * New: Windows 2008 Server compatibility. * CHANGE: Some performance optimization. * FIX: SIP/ Dns can generate void extensions. * FIX: in process dn expresion, the duplicate filter deletes non duplicate entries. download: http://sourceforge.net/projects/activa/files/ doc: http://activa.sourceforge.net/readme.html many thanks to Activa Team --- Marek Cervenka === Congratulations for this release !! I'm not sure this list is the right place to ask but is it possible, to install ActivaTSP on a single Windows server and simply configure each Windows XP/Vista station's TAPI stack to use this server resource (avoiding adding any software on client PCs) ? ___ -- 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] Calls Dropping
The info you need is here http://www.voip-info.org/wiki/view/Asterisk+config+logger.conf Ish Dan Journo wrote: Hello, We have a problem that calls seem to be dropping for no reason. Is there any way to write a debug log to disk so that I can check it as soon as a call is lost? It happens randomly once or twice a day to different callers. Many thanks Dan ___ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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] Calls Dropping
On 11 Dec 2009, at 11:19, Dan Journo wrote: Is there any way to write a debug log to disk so that I can check it as soon as a call is lost? It happens randomly once or twice a day to different callers. /var/log/asterisk/full? Most 'standard' setups produce it. Failing that google will reveal how to do this. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Dropping
Thanks. I didnt stop that. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: 11 December 2009 11:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Dropping The info you need is here http://www.voip-info.org/wiki/view/Asterisk+config+logger.conf Ish Dan Journo wrote: Hello, We have a problem that calls seem to be dropping for no reason. Is there any way to write a debug log to disk so that I can check it as soon as a call is lost? It happens randomly once or twice a day to different callers. Many thanks Dan ___ -- 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 -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 ___ -- 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] VUC Dec 11 @ 12 Noon EST: g729 transcoding, software hardware
Hi, We had a last-minute cancellation from Vivox for today's conference. It happens that someone suggested a guest idea, Howler Technologies CTO Jay Fenton, who agreed to join the call from the road. Anything you want to know about transcoding to and from g729 is out topic for the first hour. My pal David Duffet knows this technology well and has kindly signed in to help guide us through this as well. Just before this time in your local time zone : http://vuc.me/next (12 Noon EST) why not join us on IRC: #vuc on Freenode.net anytime SIP:200...@login.zipdx.com Skype:vuc.me (or skypeld.vuc.me for reduced bandwidth) PSTN: (567) 252-2286 Java web widget: http://vuc.me/call If you have a shipping address in North America, you can vie for a free Polycom ip335 : http://bit.ly/8px6al Independent of your location in the world, Howler Technologies is offering some free licenses for their g729 transcoding technology. See you there! /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Unregisteres IAX Friend Randomly
Hi, I've been having a strange problem recently where real-time asterisk will unregister a IAX friend at random times when the registration should not have expired. I have a Zoiper soft phone client (on windows) connecting to asterisk over a LAN (no firewalls). The default reregister time of 60 seconds is used, but the asterisk server unregisters the client (sets regseconds to 0 in the database) after a seemingly random time after registration say, 15 seconds (which is 45 seconds before registration expiry) I set iax2 debug on, and set the core debug level to 5 so I could see the IAX2 control frames on the console. I use pgAdmin-III to watch the value of regseconds in the database change from a registered value to 0. When the unregister happens, there are no frames sent to / received from the client, and nothing else on the asterisk console. It just seems like asterisk decided to unregister the client for no reason. At this point placing a call to the client will fail, until the client reregisters (at the correct time) 45 seconds later. I have seen this happen on both 1.6.2-rc8 and a trunk asterisk from yesterday (on different machines). I'm currently installing 1.6.0 to test that as well. This only seems to happen with real-time asterisk. (I'm using Postgres for the backend database and the pgsql driver in extconfig.conf) Any ideas what's going on here? Is this a known issue? Thanks in advance. Regards, Dr. Nic Colledge ___ -- 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] max. no. of conferences supported
What are the limits with asterisk server running on one decent (4GB, 4 CPU etc.) machine. There are a LOT of factors involved. You will likely have to do your own testing with just the specific features you want. How many MeetMe conferences it can support? What is the limit of number of participants per conference? Are you doing any transcoding? What technologies are the participants using (dahdi, sip, iax, etc)? If you're doing a conference with only sip participants and no transcoding, on the hardware you mention you should be able to comfortably host a conference with 100 participants, possibly more. I can't help you out with specific numbers, as none of the systems I administer do conferences larger than this. As for the number of conferences, I've seen one system with similar hardware specs regularly host a dozen conferences without issue. Most of these conferences have between 5 and 10 participants. Is it possible to support 1000 users in Asterisk? What is the kind of hardware needed for this? Yes, there are a good number of asterisk installations with more than 1000 users. In a recent interview with Mark Spencer, he mentioned an installation with 150,000 users. What kind of hardware all depends on what features/services you want to provide. All I can say is that you should pick and choose features/services carefully if you intend to have a lot of users. By default, asterisk will enable everything. Change it to only enable what you need. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.1.11 Fax
Steve Underwood wrote: Something is wrong if Asterisk is sending: a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG Spandsp supports T38FaxFillBitRemoval, but neither spandsp or Commetrex support the other two options. The Commetrex guys have said so in the FoIP working group. Agreed... if he's actually running a FAX application on Asterisk. If this is a bridged call, the other endpoint may have offered to support those features. Without a description of the actual call scenario, we can only guess :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] zttool don't show NT mode with OctoBRI
Hi, Using Xorcom's bristuff-0.4.0-RC4-xr7.tar.gz (ie asterisk 1.4.25) with an old Junghanns OctoBRI (ie not the 2.0 version), zttool shows every port in TE mode, though half of them are in NT. Alarm column shows OK for every port, such as : OK octoBRI PCI ISDN Card 1 Span 1 [TE] Lay Globally, this system doesn't work (can't dial out through zaptel) Replacing this board with a Junghanns QuadBRI PCIe card, zttool shows correct TE/NT modes but Alarm column shows OK for every port,such as : OK quadBRI PCI ISDN Card 1 Span 3 [NT] (ca Globally, this system does work (can dial out through zaptel) When I replaced the OctoBRI with the QuadBRI, I didn't change anything in Zaptel/Asterisk config (I simply turned the PC off and on) except removing unuseful lines in /etc/zaptel.conf and /etc/asterisk/zapata.conf. Here are those 2 files: # cat zapata.conf [channels] language=fr context=remote pridialplan=unknown prilocaldialplan=unknown internationalprefix = 00 nationalprefix = 0 usecallerid=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes ;;;echocancel=yes ;;;echocancelwhenbridged=yes ;echotraining=yes ;echotraining=800 ;relaxdtmf=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no ;callprogress=yes ;progzone=us ;signalling=bri_cpe_ptmp group = 1 switchtype=euroisdn signalling=bri_cpe channel = 1-2 echocanceller=oslec,1-2 group = 1 switchtype=euroisdn signalling=bri_cpe channel = 4-5 echocanceller=oslec,4-5 group = 1 switchtype=euroisdn signalling=bri_net channel = 7-8 echocanceller=oslec,7-8 group = 1 switchtype=euroisdn signalling=bri_net channel = 10-11 echocanceller=oslec,10-11 # cat ../zaptel.conf loadzone=fr defaultzone=fr # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,2,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 File /etc/default/zaptel simply includes MODULES=qozap. Any hint ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Free Fax for Asterisk
Can I install my free fax for asterisk license on more than one machine? I.e using my digiun account to download the free FFA module, am I restricted to just the first machine I put it on, or can I put the free FFA on multiple servers? Thanks, --Warren Selby ___ -- 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] question on register
Where in the code does something like: register = user[:secret[:authuse...@host[:port][/extension] from sip.conf 1) get parsed 2) actually register. I tried looking in channels/chan_sip.c and don't see where that happens. Can someone point me the right file and or function. Thanks, Jerry ___ -- 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] Echo issue
The echo between our extensions (using Polycom 550 handsets) disappears once I removed the Digium echo module. Are you routing internal calls from SIP - DAHDI - SIP? The digium echo module will not have any effect on pure SIP - SIP calls. Do you have acoustic echo cancellation active on the Polycom phones? What kind of settings do you recommend for the txgain and rxgain? Ideally, you will need to measure to find out what settings you want. See this page on the wiki (see the note on values for PRI circuits): http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment (use dahdi_monitor instead of ztmonitor) You can also just experiment with different values. Change just one setting at a time, and then reload Dahdi. Try this to start: txgain = 0.0 rxgain = 1.0 and then on the asterisk cli, enter: module reload chan_dahdi.so If that doesn't help, try increasing to rxgain=2.0. Keep going until it sounds better. You may want to try negative values for txgain. Do I make the gain changes in chan_dahdi.conf? Yes. Make sure to put them before your channel numbers. You can specify values on a per-channel basis. This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Did you use these same settings when you were using the hardware echo module? - Noah ___ -- 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] Dahdi and Junghanns QuadBRI
Hi, Using this OctoBRI card in a bristuff-0.4.0-RC4-xr7.tar.gz-enabled machine, I discovered zttool was not able to detect its NT mode. I opened a thread on this as I'm suspecting this older type of OctoBRI card might need a specific driver. Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple sip trunks
Thanks - have done that and am now trying a one out. However, I'd still like to know whether 1 asterisk server can support multiple trunks/registry entries. Does it cause problems? Thanks John 2009/12/3 Tim Nelson tnel...@rockbochs.com: - John Taylor j...@vetsurgeon.org.uk wrote: I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John Why not go with a real carrier that can send you proper DID and DNIS information for each call? Rather than trying to configure/code/etc around the problem with the ITSP, use an ITSP that does things correctly. There are many people here on asterisk-users that can recommend a proper ITSP. If you want pure business response, head over to asterisk-biz and ask there. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- 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] Free Fax for Asterisk
Warren Selby wrote: Can I install my free fax for asterisk license on more than one machine? I.e using my digiun account to download the free FFA module, am I restricted to just the first machine I put it on, or can I put the free FFA on multiple servers? I would believe you get a single license, and that license is good for a single instance on a single machine. Installing it to another machine would require you to transfer it, making the other machine no longer valid to use it on. Otherwise you'd see people installing 100s of VMs and installing a single FFA license on all of them :) Leif Madsen. ___ -- 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] Free Fax for Asterisk
On Fri, Dec 11, 2009 at 10:30 AM, Leif Madsen leif.mad...@asteriskdocs.orgwrote: Warren Selby wrote: Can I install my free fax for asterisk license on more than one machine? I.e using my digiun account to download the free FFA module, am I restricted to just the first machine I put it on, or can I put the free FFA on multiple servers? I would believe you get a single license, and that license is good for a single instance on a single machine. Installing it to another machine would require you to transfer it, making the other machine no longer valid to use it on. Otherwise you'd see people installing 100s of VMs and installing a single FFA license on all of them :) Leif Madsen. ___ -- 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 That's actually what I thought as well, until I read the description on the Product page on the digium site for the Free FFA ( http://store.digium.com/productview.php?product_code=804-7): Free Fax For Asterisk is provided, one *per installation* of Asterisk, to customers without charge. The reason I ask is I've got a few different clients that want to test Fax for Asterisk. I was planning on buying the Free FFA license using my digium account, but I don't want to get locked out after installing it on my first client's machine, and I was hoping to avoid creating multiple digium.com user accounts if it was possible. Oh well, if it's not possible, it's not a big deal to create the multiple accounts, even if I'm the only one that will ever use them... -- Thanks, --Warren Selby http://www.selbytech.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple sip trunks
On Fri, Dec 11, 2009 at 10:23 AM, John Taylor j...@vetsurgeon.org.uk wrote: Thanks - have done that and am now trying a one out. However, I'd still like to know whether 1 asterisk server can support multiple trunks/registry entries. Does it cause problems? yes, Asterisk does support multiple registry entries... if it didn't ... it would be just a crippled sip endpoint lets say more ... Asterisk can do whatever you want it to do (within reason and technical boundaries); just code it in or request a feature Martin Thanks John 2009/12/3 Tim Nelson tnel...@rockbochs.com: - John Taylor j...@vetsurgeon.org.uk wrote: I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John Why not go with a real carrier that can send you proper DID and DNIS information for each call? Rather than trying to configure/code/etc around the problem with the ITSP, use an ITSP that does things correctly. There are many people here on asterisk-users that can recommend a proper ITSP. If you want pure business response, head over to asterisk-biz and ask there. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
The echo between our extensions (using Polycom 550 handsets) disappears once I removed the Digium echo module. Are you routing internal calls from SIP - DAHDI - SIP? The digium echo module will not have any effect on pure SIP - SIP calls. Do you have acoustic echo cancellation active on the Polycom phones? Internal calls should be SIP to SIP. Yes we do have the acoustic echo cancellation active on the Polycom phones. This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Did you use these same settings when you were using the hardware echo module? Yes, I believe so. I asked an Asterisk expert to make sure everything is working correctly when installing the hardware module. If the setting don't look correct, what should be there when we use the hardware module? Thank you! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime Database Tables
I'm actually there, but I was wondering if the tables there are up to date and if any changes took place. I see all kinds of comments about changes. You could go ahead and install and then look at the table structure using your dbms. - Noah ___ -- 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] chan_dahdi.conf for TDM404E
Hi there, I am trying to configure chan_dahdi.conf for TDM404E. Should I separate channels for dialing out and recieveing calls on this card or should I leave it random so that outgoing and incoming call get first available channels. ;FXO Modules group = 2 echocancel = yes signalling = fxs_ks context = Incoming Is it possible to define more than one context here as above mentioned config is serving only context incoming. Please help in this regard. Thanks Shahnawaz ___ -- 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] ATA FXO
I'm looking for a reliable ATA FXO/FXS adapter. Linksys 3102 - a lot of echo problem + two of them died within a year (not reliable) Sangoma USBFXO - problem installing drive in Gentoo. I've tried two Chines units: AG-188N and YGW30B none are of them have real FXO port that will register with Asterisk. Any other recommendations; (I don't like internal cards). -- Joseph ___ -- 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] multiple sip trunks
I assume if all the SIP trunks are to the same host/port, Asterisk cannot distinguish which trunk is active when an incoming call is made- it will dump all incoming calls to the context specified in the last trunk entry of sip.conf Thanks John 2009/12/11 Martin asteriskl...@callthem.info: On Fri, Dec 11, 2009 at 10:23 AM, John Taylor j...@vetsurgeon.org.uk wrote: Thanks - have done that and am now trying a one out. However, I'd still like to know whether 1 asterisk server can support multiple trunks/registry entries. Does it cause problems? yes, Asterisk does support multiple registry entries... if it didn't ... it would be just a crippled sip endpoint lets say more ... Asterisk can do whatever you want it to do (within reason and technical boundaries); just code it in or request a feature Martin Thanks John 2009/12/3 Tim Nelson tnel...@rockbochs.com: - John Taylor j...@vetsurgeon.org.uk wrote: I want to use an asterisk box to provide a voip service to a number of separate companies. I have a VOIP provider who I want to trunk with. As far as I can see it there are 2 options 1. Have 1 SIP trunk to one account at the provider who gives me multiple incoming numbers; this is less than optimal as the provider does not provide the DID number in the sip header; I only get the account number. I have the option to set called line presentation but this will stop CLID 2. Have multiple sip trunks to multiple accounts at the provider. Is this an advisable thing to do? I notice asterisk does not handle the incoming context correctly (all incoming calls go to the last incoming context defined in sip.conf), but I can extract the account called via the EXTEN variable. I would be looking at providing around 20 companies with accounts (all very small), and would prefer option (2) to enable failover to a number they specify. Thanks for any light shed John Why not go with a real carrier that can send you proper DID and DNIS information for each call? Rather than trying to configure/code/etc around the problem with the ITSP, use an ITSP that does things correctly. There are many people here on asterisk-users that can recommend a proper ITSP. If you want pure business response, head over to asterisk-biz and ask there. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- 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] G729 Pass through
Hi; I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my endpoints. It seems that when I enable G729 on my peers in sip.conf and make a call I am getting the following errors: Called crp_uk/806575011971553141421 Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec translation path from g729 to ulaw Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Both my end points (Aastra phone) and my sip carrier support G729, so this should be simple pass-through. Snippet of my peer crp_uk: [crp_uk] disallow=all allow=ulaw allow=alaw allow=g729 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Pass through
Have you paied for and imported g729 licenses from digium so that asterisks can use g729? http://store.digium.com/productview.php?category_id=5product_code=8G729 CODEC James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information =is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey Forman Sent: Friday, December 11, 2009 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] G729 Pass through Importance: High Hi; I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my endpoints. It seems that when I enable G729 on my peers in sip.conf and make a call I am getting the following errors: Called crp_uk/806575011971553141421 Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec translation path from g729 to ulaw Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Both my end points (Aastra phone) and my sip carrier support G729, so this should be simple pass-through. Snippet of my peer crp_uk: [crp_uk] disallow=all allow=ulaw allow=alaw allow=g729 image001.jpg___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Pass through
I have – but I don’t see why that would be required for pass – through? The codec purchase should only be required if I wanted to leave voicemail in G729 or MOH. If my end points support G729 and I am advertising it in the invite, and negotiating it with the 200OK, I don’t see why its not allowing pass through. --Dovey -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *James A. Shigley *Sent:* Friday, December 11, 2009 1:16 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] G729 Pass through Have you paied for and imported g729 licenses from digium so that asterisks can use g729? http://store.digium.com/productview.php?category_id=5product_code=8G729CODEC James Shigley *Monroe** Telephone Answering Service* 409-981-9213** Infinity 5.51,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which may be confidential or privileged. The information =is intended to be for the use of the individual or entity named above. If you are not the intended recipient, be aware that any disclosure, copying, distribution or use of the contents of this information is prohibited. If you have received this email in error, please notify the sender immediately by reply to sender only message and destroy all electronic and hard copies of the communication, including attachments. [image: cid:image003.png@01C9F268.65A4F5C0] *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dovey Forman *Sent:* Friday, December 11, 2009 12:06 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] G729 Pass through *Importance:* High Hi; I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my endpoints. It seems that when I enable G729 on my peers in sip.conf and make a call I am getting the following errors: Called crp_uk/806575011971553141421 Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec translation path from g729 to ulaw Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Both my end points (Aastra phone) and my sip carrier support G729, so this should be simple pass-through. Snippet of my peer crp_uk: [crp_uk] disallow=all allow=ulaw allow=alaw allow=g729 image001.jpg___ -- 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] multiple sip trunks
I assume if all the SIP trunks are to the same host/port, Asterisk cannot distinguish which trunk is active when an incoming call is made- it will dump all incoming calls to the context specified in the last trunk entry of sip.conf No. SIP uses authentication (well, I guess you can not use authentication). Asterisk (and almost any SIP gateway) will correctly match the call to the trunk based on the authentication. Even if you didn't send any authentication info, asterisk will try to match the call as a guest call. It is common practice to not allow unauthenticated SIP traffic. - Noah ___ -- 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] ATA FXO
Grandstream HT503 On Fri, 2009-12-11 at 10:37 -0700, Joseph wrote: I'm looking for a reliable ATA FXO/FXS adapter. Linksys 3102 - a lot of echo problem + two of them died within a year (not reliable) Sangoma USBFXO - problem installing drive in Gentoo. I've tried two Chines units: AG-188N and YGW30B none are of them have real FXO port that will register with Asterisk. Any other recommendations; (I don't like internal cards). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Pass through
Hi! Are you sure you are getting Astrisk out of the media path? I guess reinvite must be allowed. Then it should work without transcoding licenses. Maybe you should take a look at the SIP DEBUG info to see what codec Asterisk is trying to negotiate with the trunk. You could disallow alaw and ulaw for a test. Christian 2009/12/11 Dovey Forman dovey.for...@idt.net: Hi; I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my endpoints. It seems that when I enable G729 on my peers in sip.conf and make a call I am getting the following errors: Called crp_uk/806575011971553141421 Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec translation path from g729 to ulaw Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) Both my end points (Aastra phone) and my sip carrier support G729, so this should be simple pass-through. Snippet of my peer crp_uk: [crp_uk] disallow=all allow=ulaw allow=alaw allow=g729 ___ -- 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] ATA FXO
I'll check it out, but Grandstream HT503 doesn't have a good introduction on voip-wiki web-page: http://www.voip-info.org/wiki/view/HT-503 -- Joseph On 12/11/09 19:37, jonas kellens wrote: Grandstream HT503 On Fri, 2009-12-11 at 10:37 -0700, Joseph wrote: I'm looking for a reliable ATA FXO/FXS adapter. Linksys 3102 - a lot of echo problem + two of them died within a year (not reliable) Sangoma USBFXO - problem installing drive in Gentoo. I've tried two Chines units: AG-188N and YGW30B none are of them have real FXO port that will register with Asterisk. Any other recommendations; (I don't like internal cards). ___ -- 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] ATA FXO
-Original Message- From: Joseph [mailto:syscon...@gmail.com] Sent: Friday, December 11, 2009 11:37 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ATA FXO I'm looking for a reliable ATA FXO/FXS adapter. Linksys 3102 - a lot of echo problem + two of them died within a year (not reliable) Sangoma USBFXO - problem installing drive in Gentoo. I've tried two Chines units: AG-188N and YGW30B none are of them have real FXO port that will register with Asterisk. Any other recommendations; (I don't like internal cards). -- Joseph We have had good luck with the Mediatrix gateways. -Connor ___ -- 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] ATA FXO
On 12/11/09 12:52, Connor Spiess wrote: -Original Message- From: Joseph [mailto:syscon...@gmail.com] Sent: Friday, December 11, 2009 11:37 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ATA FXO I'm looking for a reliable ATA FXO/FXS adapter. Linksys 3102 - a lot of echo problem + two of them died within a year (not reliable) Sangoma USBFXO - problem installing drive in Gentoo. I've tried two Chines units: AG-188N and YGW30B none are of them have real FXO port that will register with Asterisk. Any other recommendations; (I don't like internal cards). -- Joseph We have had good luck with the Mediatrix gateways. -Connor Looks very interesting, but they are all either FXS or FXO; it would be practical if they could make it 2xFX0 and or 2xFXS or modular design. Cisco had such unit but they discontinued it. -- Joseph ___ -- 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] ATA FXO
-Original Message- From: Joseph [mailto:syscon...@gmail.com] Sent: Friday, December 11, 2009 1:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ATA FXO On 12/11/09 12:52, Connor Spiess wrote: -Original Message- From: Joseph [mailto:syscon...@gmail.com] Sent: Friday, December 11, 2009 11:37 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ATA FXO I'm looking for a reliable ATA FXO/FXS adapter. Linksys 3102 - a lot of echo problem + two of them died within a year (not reliable) Sangoma USBFXO - problem installing drive in Gentoo. I've tried two Chines units: AG-188N and YGW30B none are of them have real FXO port that will register with Asterisk. Any other recommendations; (I don't like internal cards). -- Joseph We have had good luck with the Mediatrix gateways. -Connor Looks very interesting, but they are all either FXS or FXO; it would be practical if they could make it 2xFX0 and or 2xFXS or modular design. Cisco had such unit but they discontinued it. -- Joseph You could also check out the Audio Codes gateways if the Grandstream doesn't work out for you. They make FXO/FXS gateways. They were reliable boxes for us but this was to a non-asterisk PBX over MGCP. I mention them cause I know they make a SIP based one. Hope your grandstream works for you so you won't have to go down this path. Goodluck. --Connor ___ -- 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] chan_dahdi.conf for TDM404E
On Fri, Dec 11, 2009 at 10:25:33AM -0700, mir shahnawaz wrote: Hi there, I am trying to configure chan_dahdi.conf for TDM404E. Should I separate channels for dialing out and recieveing calls on this card or should I leave it random so that outgoing and incoming call get first available channels. ;FXO Modules group = 2 echocancel = yes signalling = fxs_ks context = Incoming Is it possible to define more than one context here as above mentioned config is serving only context incoming. Please help in this regard. The context is set for each channel . So you could set it e.g.: context = incoming channel = 1-3 context = incoming-4 channel = 4 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
On Wed, 9 Dec 2009, Michelle Dupuis wrote: However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Did you find a solution -- inquiring minds want to know... -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA FXO
On 12/11/09 14:44, Connor Spiess wrote: Looks very interesting, but they are all either FXS or FXO; it would be practical if they could make it 2xFX0 and or 2xFXS or modular design. Cisco had such unit but they discontinued it. -- Joseph You could also check out the Audio Codes gateways if the Grandstream doesn't work out for you. They make FXO/FXS gateways. They were reliable boxes for us but this was to a non-asterisk PBX over MGCP. I mention them cause I know they make a SIP based one. Hope your grandstream works for you so you won't have to go down this path. Goodluck. --Connor I think I'll try first Zoom 5801, not sure if it is a perfect solution but based on some other reviews this unit has no echo on PSTN line. -- Joseph ___ -- 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] Can't restart asterisk from script
Looks like single quotes did the trick. No idea why...but the error is gone from my log -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, December 11, 2009 4:23 PM To: Asterisk Users List Subject: Re: [asterisk-users] Can't restart asterisk from script On Wed, 9 Dec 2009, Michelle Dupuis wrote: However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Did you find a solution -- inquiring minds want to know... -- 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 -- 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] Terminate T.38 to PSTN
Hi, Has terminating T.38 to PSTN found its way into the asterisk 1.6 mainline yet? I remember seeing an app_gateway floating around at some point a while ago, but I never had any luck with it. Thanks. -- James ___ -- 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] ATA FXO
On Fri, Dec 11, 2009 at 12:44 PM, Connor Spiess cspi...@idea-ma.com wrote: Joseph You could also check out the Audio Codes gateways if the Grandstream doesn't work out for you. They make FXO/FXS gateways. They were reliable boxes for us but this was to a non-asterisk PBX over MGCP. I mention them cause I know they make a SIP based one. We use AudioCodes MP-114 2FXS/2FXO and they have been rock solid. I have a bunch used for faxing connected back to Asterisk over SIP. I will say that I have had a LOT of issues with faxing on the larger GrandStream GXW-4024s and had to replace them. I put a AudioCodes MP-124 in and have had no complaints since. -Jonathan ___ -- 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] Terminate T.38 to PSTN
James Lamanna wrote: Has terminating T.38 to PSTN found its way into the asterisk 1.6 mainline yet? I remember seeing an app_gateway floating around at some point a while ago, but I never had any luck with it. It has not, no. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] multiple sip trunks
I have multiple trunks to the same ITSP. Incoming calls to any trunk go to the last incoming label defined in those trunks' contexts in sip.conf. My ITSP insists on insecure=very in the trunk context; is this the cause? John 2009/12/11 Noah Miller noahisaacmil...@gmail.com: I assume if all the SIP trunks are to the same host/port, Asterisk cannot distinguish which trunk is active when an incoming call is made- it will dump all incoming calls to the context specified in the last trunk entry of sip.conf No. SIP uses authentication (well, I guess you can not use authentication). Asterisk (and almost any SIP gateway) will correctly match the call to the trunk based on the authentication. Even if you didn't send any authentication info, asterisk will try to match the call as a guest call. It is common practice to not allow unauthenticated SIP traffic. - Noah ___ -- 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] Can't restart asterisk from script
Un-top-posting... On Wed, 9 Dec 2009, Michelle Dupuis wrote: However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) On Behalf Of Steve Edwards Sent: Friday, December 11, 2009 4:23 PM Did you find a solution -- inquiring minds want to know... On Fri, 11 Dec 2009, Michelle Dupuis wrote: Looks like single quotes did the trick. No idea why...but the error is gone from my log I believe this to be a red-herring. Single or double quotes should produce exactly the same result in your situation. -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't restart asterisk from script
restart when convenient -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, December 11, 2009 4:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't restart asterisk from script Un-top-posting... On Wed, 9 Dec 2009, Michelle Dupuis wrote: However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) On Behalf Of Steve Edwards Sent: Friday, December 11, 2009 4:23 PM Did you find a solution -- inquiring minds want to know... On Fri, 11 Dec 2009, Michelle Dupuis wrote: Looks like single quotes did the trick. No idea why...but the error is gone from my log I believe this to be a red-herring. Single or double quotes should produce exactly the same result in your situation. -- 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 -- 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] Playing a message if my call lands in their voicemail
Hi All, My client makes manual sales calls to prospects. He is often sent to voicemail on the prospect's side. If he finds himself having to leave a message, he would like to be able to press a key and let a pre-recorded message play into the prospect's vmail box. This is so he can maintain consistency in his message. Can anyone offer suggestions of how I could accomplish this functionality? I am running the latest and greatest Asterisk. thanks in advance. JR ___ -- 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] T38 Passthrough 1.6.1.12-rc1 Good Results
Hi All, I've been knee deep in T38 faxing for a couple of weeks now, trying to find a version of Asterisk that would pass through T38 with an Audiocodes Mediant 1000 and MP203 ATA. I had problems with 1.6.0.x through 1.6.1.10. Tested 6 different versions. Either it just would not work or fail back to G.711, or re-invite with wrong T38FaxMaxDatagram sizes, faxes would work one-way and not the other, and so on, issue after issue. After trading some emails with the dev list, I learned a bit more about what a good T38 negotiation should look like which helped out quite a bit. I started reading the changelog for the various versions of 1.6.0.x and 1.6.1.x, focusing on chan_sip updates and fixes. There have been a lot of updates regarding T38, tweaks, patches, adding functionality, and there was a total re-write of the stack as well. All this within the past few months. Just two days ago 1.6.1.12-rc1 was uploaded, changelog noted a handful of more T38 changes. A particularly interesting one for me was: 2009-11-30 21:55 + [r231694] Kevin P. Fleming kpflem...@digium.com I was getting T38FaxMaxDatagram size miss-matches in the T38 negotiation which was causing failures, IFP byte miss-match and buffer overflow errors. This update has resolved these particular issues and with my specific lab testing, T38 faxing is negotiating faster and completing quicker. So just to check myself, I defaulted my lab setup and rebuilt with just basic configs on the Mediant 1000, the MP203 (behind a NAT) and Asterisk, sent several faxes coming and going with no errors, all T38 negotiated faxes nailed up at 14400. I'll be doing a lot more testing next week, but I'm very happy with the results so far. I wanted to share so you all were aware of the progress that is being made in this particular area and also thank the dev team for responding to the bug tracker, taking suggestions for improvements and doing the coding to make Asterisk the best it can be. I can't wait for T38 gateway. Keep up the good work. Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] ATA FXO
On 12/11/09 14:05, Jonathan Thurman wrote: On Fri, Dec 11, 2009 at 12:44 PM, Connor Spiess cspi...@idea-ma.com wrote: Joseph You could also check out the Audio Codes gateways if the Grandstream doesn't work out for you. They make FXO/FXS gateways. They were reliable boxes for us but this was to a non-asterisk PBX over MGCP. I mention them cause I know they make a SIP based one. We use AudioCodes MP-114 2FXS/2FXO and they have been rock solid. I have a bunch used for faxing connected back to Asterisk over SIP. I will say that I have had a LOT of issues with faxing on the larger GrandStream GXW-4024s and had to replace them. I put a AudioCodes MP-124 in and have had no complaints since. -Jonathan Thank for suggestion. Well, it is not that cheap but the problem with their equipment is luck support and decent manual. Whatever I google about AudioCodecs everybody seems to be straggling with the setup; I don't think this should be that hard to write a decent instructions if they want to sell their product. Maybe they have a good product but without support it will not mean much. eg.: http://www.trixbox.org/forums/trixbox-forums/trunks/trixbox-2-2-and-audio-codes-mp-114-fxo-setup -- Joseph ___ -- 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] ATA FXO
On Fri, 11 Dec 2009, Joseph wrote: On 12/11/09 14:05, Jonathan Thurman wrote: On Fri, Dec 11, 2009 at 12:44 PM, Connor Spiess cspi...@idea-ma.com wrote: Joseph You could also check out the Audio Codes gateways if the Grandstream doesn't work out for you. They make FXO/FXS gateways. They were reliable boxes for us but this was to a non-asterisk PBX over MGCP. I mention them cause I know they make a SIP based one. We use AudioCodes MP-114 2FXS/2FXO and they have been rock solid. I have a bunch used for faxing connected back to Asterisk over SIP. I will say that I have had a LOT of issues with faxing on the larger GrandStream GXW-4024s and had to replace them. I put a AudioCodes MP-124 in and have had no complaints since. -Jonathan Thank for suggestion. Well, it is not that cheap but the problem with their equipment is luck support and decent manual. Whatever I google about AudioCodecs everybody seems to be straggling with the setup; I don't think this should be that hard to write a decent instructions if they want to sell their product. Maybe they have a good product but without support it will not mean much. eg.: http://www.trixbox.org/forums/trixbox-forums/trunks/trixbox-2-2-and-audio-codes-mp-114-fxo-setup Well now that you have shot down just about every decent piece of hardware that has been suggested, you are probably left with designing your own! I totally disagree with your comments on Audiocodes... excellent product. j ___ -- 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] ATA FXO
On 12/12/09 04:02, Jeff LaCoursiere wrote: [snip] Thank for suggestion. Well, it is not that cheap but the problem with their equipment is luck support and decent manual. Whatever I google about AudioCodecs everybody seems to be straggling with the setup; I don't think this should be that hard to write a decent instructions if they want to sell their product. Maybe they have a good product but without support it will not mean much. eg.: http://www.trixbox.org/forums/trixbox-forums/trunks/trixbox-2-2-and-audio-codes-mp-114-fxo-setup Well now that you have shot down just about every decent piece of hardware that has been suggested, you are probably left with designing your own! I totally disagree with your comments on Audiocodes... excellent product. j I've not totally excluded Audiocodes yet; but I don't want to keep hunting Internet for hours to try to configure simple things; I've notice some folks keep struggling to configure the Audiocodes unit to pass caller ID from PSTN to asterisk and I've not found a solution yet. I already have enough door stoppers and I don't need another one especially that this one is not cheap. I don't mind to pay the price but I don't want to spend hours or days trying to configure it. For example Linksys 3102 is easy to configure plenty of support and examples; it take me below 5min to set it up (the only problem with those units is the echo on PSTN line). I would like to buy brand new original Sipura 3000 but cannot find them anymore, I have been using two for over 4-years and they are working perfectly for me. -- Joseph ___ -- 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] ATA FXO
On Fri, Dec 11, 2009 at 7:52 PM, Joseph syscon...@gmail.com wrote: [snip] Thank for suggestion. Well, it is not that cheap but the problem with their equipment is luck support and decent manual. I actually find the Quick-start guide that comes in the box the most useful, if you aren't doing anything strange. Whatever I google about AudioCodecs everybody seems to be straggling with the setup; I don't think this should be that hard to write a decent instructions if they want to sell their product. Maybe they have a good product but without support it will not mean much. While I agree about the manual being a little difficult, the actual support from AudioCodes is great. They want you to get support through the reseller or distributor, but you can purchase direct support too. If you do that then you can call them up and talk to an engineer. They will even Web-X in and show you how to do something if they don't have a quick how-to document to email you on the subject. The interface is also a bit overwhelming at first, and forget the console. However once you get the configuration set, export it out as a text file and make a template. I can't speak specifically to Caller-ID on FXO ports, as I mainly use them for FXS and local 911 gateways. -Jonathan ___ -- 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] ATA FXO
On 12/11/09 21:21, Jonathan Thurman wrote: On Fri, Dec 11, 2009 at 7:52 PM, Joseph syscon...@gmail.com wrote: [snip] Thank for suggestion. Well, it is not that cheap but the problem with their equipment is luck support and decent manual. I actually find the Quick-start guide that comes in the box the most useful, if you aren't doing anything strange. I don't want to do anything out of the ordinary. All calls are controlled by Asterisk, so call comes IN on PSTN line (3sec delay to pass caller ID) and is forwarded to Asterisk. All internal extension whatever equipment I use (sipura, linksys, iaxy101) should be able to call out on PSTN line using Asterisk dial plan. Nothing special. Whatever I google about AudioCodecs everybody seems to be straggling with the setup; I don't think this should be that hard to write a decent instructions if they want to sell their product. Maybe they have a good product but without support it will not mean much. While I agree about the manual being a little difficult, the actual support from AudioCodes is great. They want you to get support through the reseller or distributor, but you can purchase direct support too. If you do that then you can call them up and talk to an engineer. They will even Web-X in and show you how to do something if they don't have a quick how-to document to email you on the subject. The interface is also a bit overwhelming at first, and forget the console. However once you get the configuration set, export it out as a text file and make a template. I can't speak specifically to Caller-ID on FXO ports, as I mainly use them for FXS and local 911 gateways. -Jonathan I'll keep looking into it but from my experience support from equipment sellers is not the best one; all they want is to make a sale, once the product is through the door you can hardly get reply from them if you have a question. Now, paying 400.00 and still subscribing to support is a bit too much. Simple configuration plan with PBX like asterisk should be posted on tier web-page. If only Sangoma had this kind of product I jump right in; they have a good web support. -- Joseph ___ -- 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
