[asterisk-users] Asterisk 1.6 and RFC4235
Does Asterisk 1.6 fully support RFC4235? Or is it the same implementation as 1.4? Thanks. -- James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Request for information about Asterisk Business Edition
Hello, For people having experienced Asterisk Business Edition, please I need some information: - First, Can ABE be installed in a Debian or Ubuntu OS 32 and 64 bit. - Second, Can ABE be installed in a newer version of Fedora like Fedora 10 or 11. - Third, opcom, a reseller of ABE in France says that there is a support for 250 additional calls. Is this really possible, because the official digium site says that the maximum number of simultaneous calls is only 240. - Fourth, If the third point is OK ie ABE can support 500 simultaneous calls, can the system configuration described below handle this number of calls (500) with the alaw or ulaw codecs. The configuration is: Hardware Information Processors 4 Model Intel(R) Xeon(R) CPU E5420 @ 2.50GHz CPU Speed 2.49 GHz Cache Size 6.00 MB System Bogomips 19954.79 PCI Devices *none * IDE Devices *none* SCSI Devices - DELL PERC 6/i (Direct-Access) - DP BACKPLANE (Enclosure) - TSSTcorp DVD-ROM TS-L333A (CD-ROM) USB Devices - Dell Computer Corp. - Cypress Semiconductor Corp. CY7C65640 USB-2.0 TetraHub - Dell Computer Corp. Hub Best regards. --- Abdelkader Mosbah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk core dumps files
Thanks Tzafrir for your answer. Because I had some problems running safe_asterisk script to restart asterisk automatically in our callcenter , I've developed a simple script that runs from a schedule task and check if asterisk is running each minute. This is not the best solution yet but it works properly when asterisk shutdown. However it not let asterisk generate core dumps files. Is there an error in this script or what I have to change to get core dumps files from this script. #!/bin/sh # #Script para levantar el asterisk automaticamente #programado por WL echo Checking if asterisk is running a=`pidof asterisk` if [ $a != ]; then echo Everything is OK, Asterisk is UP and running; else echo Asterisk Error: NOT RUNNING trying to restart it in 5 attempts!!!; for ((i=1; i=5; i+=1)); do /usr/sbin/asterisk -g b=`pidof asterisk` if [ $b != ]; then exit fi done fi G.A.G. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for wisdom - One Asterisk system - Multi-incoming trunks
Steve Edwards wrote:
On Wed, 29 Jul 2009, Myles Wakeham wrote:
I have setup an Asterisk system for my home home office.
[snip]
The cost of all these lines with analog carriers was getting ridiculous,
so I'm moving over to a SIP carrier. I created one account for a single
phone number with a SIP carrier (BroadVoice)
[snip]
I've never used BroadVoice, so I have nothing good or bad to say about
them. I've used Vitelity.net for several years and am pleased with them.
I have a nominal monthly fee, pay per minute account. They get $1.49 a
month for a DID and $0.0144 per minute. You'd have to use about 2,600
minutes (about 44 hours) before it would cost as much as a $40 per month
analog. They have an unlimited inbound for $7.95 a month.
I started the process today to get our other phone numbers moved over to
BroadVoice.
[snip]
Vitelity.net charges $18 per number ported. I've never done this.
My approach is to have one trunk provided by the SIP provider. All
numbers are allocated to that trunk (BroadVoice let me do that when I
setup the number transfer). Asterisk receives an incoming call on that
trunk and determines the calling number that it was requesting (not sure
how to get this, but Broadvoice assured me I could). Anyway after
determining what the call was destined for, I then route the call to the
appropriate context in the extensions to handle it.
The calls should be delivered with the DID (aka DNIS, DDI, etc). Usually
you pick this up as the ${EXTEN} in your dialplan and go from there.
[snip]
Broadvoice, however, won't let me change the outgoing caller ID.
Apparently they have to do this on a trunk by trunk basis. So if I want
to have an outgoing call go through line 1 (let's say its ACME Inc), I
want it to show 'XXX-XXX- Acme Inc' for the Caller ID.
[snip]
Being able to specify the caller ID number depends on the carrier.
Vitelity.net does. Specifying the caller ID name is not going to work. The
way it works (from 40,000 feet) is that the name is not passed onto the
real telephone system. The carrier for the dialed number looks up the
number in a database and presents that to the dialed number. If you dial
another VOIP account (sip:john-sm...@example.com) your caller ID name
should be passed.
Does this sound right? Should I have purchased all separate trunks up
front and then have the phone number transfer associated with the trunk
for it? Or is this only something that will affect outgoing calls, so
its not a big deal? And what about when the line is busy? How is that
handled? I was on the phone yesterday when another call came in, and it
came in, jumped to a different extension and then eventually went to
voice mail as I didn't answer it. Will my plan to use one trunk for all
incoming lines make sense here, or am I likely to get all of this mixed
up with calls coming in for one business and being routed to the wrong
place?
I'm more comfortable with the word account than trunk. You can have
multiple DIDs numbers associated with the same account. Some providers
make you specify (via their web site) where you want the calls to go. Some
make you configure your Asterisk server so it registers with their
server. I prefer registration because it let's me change things around
easier.
I had this issue with Teliax. Basically with SIP, Teliax could not (or
the protocol won't let you) set your outbound caller ID via Asterisk.
Caller ID is set on a per account basis with Teliax when using SIP(IAX
was not working well for me with Teliax). So I have two outbound pay per
minute accounts with them. One for our home use and one for my business.
I use 51 prefix for home outbound calls and 52 prefix for business
outbound calls. Then my dialplan selects the proper account at Teliax
and you get the proper caller id set.
My inbound is still pots lines from the telco, btw. There is no
significant cost savings on inbound for telco vs VoIP here.
Lyle Giese
LCR Computer Services, Inc.
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[asterisk-users] Sound through NAT issue
Hello everyone, I'm having a hard time configuring my router to forward asterisk traffic correctly. I have the following ports being forwarded to asterisk: 5060, 1-2 Now, I can register the accounts when outside the network and I can call every extension that is inside the network. The problem is that I can't ear anything nor can the phones inside the network phone the outside phone. Is there any port I'm forgetting to forward? Best regards, Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Res: Asterisk core dumps files
hi, the -g option is right. make sure that the system allows core files (ulimit -a). Regards -- Marcus De: Gustavo A Gonzalez ggonza...@despegar.com Para: asterisk-users@lists.digium.com Enviadas: Quinta-feira, 30 de Julho de 2009 11:17:50 Assunto: Re: [asterisk-users] Asterisk core dumps files Thanks Tzafrir for your answer. Because I had some problems running safe_asterisk script to restart asterisk automatically in our callcenter , I’ve developed a simple script that runs from a schedule task and check if asterisk is running each minute. This is not the best solution yet but it works properly when asterisk shutdown. However it not let asterisk generate core dumps files. Is there an error in this script or what I have to change to get core dumps files from this script. #!/bin/sh # #Script para levantar el asterisk automaticamente #programado por WL echo “Checking if asterisk is running” a=`pidof asterisk` if [ $a != ]; then echo Everything is OK, Asterisk is UP and running; else echo Asterisk Error: NOT RUNNING trying to restart it in 5 attempts!!!; for ((i=1; i=5; i+=1)); do /usr/sbin/asterisk -g b=`pidof asterisk` if [ $b != ]; then exit fi done fi G.A.G. Veja quais são os assuntos do momento no Yahoo! +Buscados http://br.maisbuscados.yahoo.com___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound through NAT issue
On Thu, 2009-07-30 at 16:19 +0100, Paulo Santos wrote: Hello everyone, I'm having a hard time configuring my router to forward asterisk traffic correctly. I have the following ports being forwarded to asterisk: 5060, 1-2 Now, I can register the accounts when outside the network and I can call every extension that is inside the network. The problem is that I can't ear anything nor can the phones inside the network phone the outside phone. Is there any port I'm forgetting to forward? snip What happens if you set canreinvite=no in sip.conf or the appropriate sip configuration file? - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound through NAT issue
On Thu, 30 Jul 2009, Paulo Santos wrote: Hello everyone, I'm having a hard time configuring my router to forward asterisk traffic correctly. I have the following ports being forwarded to asterisk: 5060, 1-2 Now, I can register the accounts when outside the network and I can call every extension that is inside the network. The problem is that I can't ear anything nor can the phones inside the network phone the outside phone. Is there any port I'm forgetting to forward? I don't think so, but have you tried nat=yes externip=w.x.y.z localnet=q.w.e.r/m.a.s.k in sip.conf ? (Where w.x.y.z is your external IP address, and q.w.e.r/m.a.s.k is the network and netmask of your internal network - e.g. 192.168.1.0/255.255.255.0 or whtever your LAN is using) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Possibly I don't understand sip peers
[peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 read what you've put!!! The 'allow' should be 'permit' as Jared already told you (and he should know what he's talking about). insecure=port,invite -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell Sent: 29 July 2009 23:34 To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Possibly I don't understand sip peers Jared Smith wrote: On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote: I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a 404. shouldn't I be able to put in a kind of wildcard for his IP block or am I just being silly? If not, what am I doing wrong? I think you've got your syntax wrong there... permit and deny statements are used to create Access Control Lists and to limit the IP address ranges. The allow and disallow statements are to allow or disallow various codecs. They way you've specified it above, you're allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably isn't what you want. I have the codec permissions in the columns allow and disallow. Those seem to work ok. it's permit/deny/mask I seem to be having a problem with. Like I say, I don't think I understand their use or perhaps they don't work in realtime ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dialplan SIP call back problem
Hello all, I am quite new in asterisk and I am trying to create a dialplan that executes the following steps: 1. A SIP friend dials 102 extension. 2. Asterisk PBX responds with some beeps. 3. The sip friend hangs up the phone. 4. Asterisk PBX calls back to the sip friend after 30 seconds with the application music on hold. I tried to implement this using h extension but I got the following message: Spawn extension (internal, h, 1) exited non-zero on 'SIP/bt100-083b8e60' I used also dial local but the result was the same. :( I suppose that when I pick up the phone asterisk creates a thread and when I hang up the thread stops. Am I thinking correctly?? Is there any way of executing those steps only using the dialplan?? Thanks in advance. Best regards. --- Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for wisdom - One Asterisk system - Multi-incoming trunks
Jeff LaCoursiere wrote: You don't have to send the traffic back to broadvoice for outbound if you don't want or need to. Perhaps you can send the home traffic to Broadvoice and pick another carrier to send your other outbound traffic to, perhaps one that won't be so picky about your outbound CID. Thank you for this. Yes, I wasn't thinking that way at all. I suspect that I need to find a carrier that will let me have better control over CID than BroadVoice. Do you have any suggestions? I'm in Phoenix, Arizona so one that has decent network speed near us would be best. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for wisdom - One Asterisk system - Multi-incoming trunks
Lyle wrote: I had this issue with Teliax. Basically with SIP, Teliax could not (or the protocol won't let you) set your outbound caller ID via Asterisk. Caller ID is set on a per account basis with Teliax when using SIP(IAX was not working well for me with Teliax). So I have two outbound pay per minute accounts with them. One for our home use and one for my business. I use 51 prefix for home outbound calls and 52 prefix for business outbound calls. Then my dialplan selects the proper account at Teliax and you get the proper caller id set. Yes, this is the same behavior I'm seeing with Broadvoice. What seems to make it even worse for this is that when callers receive my outgoing call, its showing the correct CID for the outgoing line, but the Name that is showing is always 'BroadVoice'. I asked them to have this changed to my company name, but it doesn't seem to have had any affect. I suspect that there is some master database somewhere that recipient phones lookup based on the number to get the name? If so, its not correctly identifying our company name on phones so I'm looking for alternative outbound carriers for this. The inbound, however, works well. I'm sure its not the least expensive option out there but so far its been pretty good. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplan SIP call back problem
Alexandre Rodrigues escribió: Hello all, I am quite new in asterisk and I am trying to create a dialplan that executes the following steps: 1. A SIP friend dials 102 extension. 2. Asterisk PBX responds with some beeps. 3. The sip friend hangs up the phone. 4. Asterisk PBX calls back to the sip friend after 30 seconds with the application music on hold. I tried to implement this using h extension but I got the following message: Spawn extension (internal, h, 1) exited non-zero on 'SIP/bt100-083b8e60' I used also dial local but the result was the same. :( I suppose that when I pick up the phone asterisk creates a thread and when I hang up the thread stops. Am I thinking correctly?? Is there any way of executing those steps only using the dialplan?? Thanks in advance. Best regards. --- Alex Yes, you cannot use the same (hungup) channel to transform it into an outbound callback call. You're right about that asterisk creates a new thread for each channel, AFAIK. So your callback solution cannot be done with dialplan only. You have to create a script that upon hangup waits the time you need and then creates a callfile to originate the callback on its own call. The same script could do the job using the AMI Originate action. This is sort of a quick answer, because I haven't had the need to develop a callback solution and I'm pretty sure there's much better solutions out there with similar concepts. Regards, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype for Asterisk: Public Beta available
I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one. Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software. This is a time-expiring beta - the software will stop working on August 31. The download is also currently time-limited - it will be available until August 7 on our website. After the 31st, you would need to have purchased a license for the SfA software (sorry, no pricing that I can give you right now - that will be a separate announcement. I'm just the community guy - I have no idea about pricing or commercial contracts or the like, so please wait until that's been announced as I will find out about the same time as you do. :-) Trial purchase page: http://store.digium.com/productview.php?product_code=804-00019 JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
I have problems with it... [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1153 skype_cp_handler: Found license 'XX' providing 1 concurrent calls [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1000 display_host: Skype For Asterisk Host-ID: X [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1320 sfa_startup: Found a total of 1 Skype For Asterisk licenses [Jul 30 14:34:21] WARNING[30613]: core.cpp:286 kill_skypewatcher: sending SIGTERM to 30614 failed with No such process *CLI [Jul 30 14:34:27] ERROR[30529]: core.cpp:1551 sfa_startup: Skype engine failed to start. [Jul 30 14:34:27] ERROR[30529]: chan_skype.c:3032 load_module: Unable to start Skype For Asterisk library. John Todd escreveu: I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one. Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software. This is a time-expiring beta - the software will stop working on August 31. The download is also currently time-limited - it will be available until August 7 on our website. After the 31st, you would need to have purchased a license for the SfA software (sorry, no pricing that I can give you right now - that will be a separate announcement. I'm just the community guy - I have no idea about pricing or commercial contracts or the like, so please wait until that's been announced as I will find out about the same time as you do. :-) Trial purchase page: http://store.digium.com/productview.php?product_code=804-00019 JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diego Aguirre (DagMoller) Infodag Consultoria FWD#: 459696 Enum#: +55 21 8871-4916 (e164.org) DUNDi-br#: 21 8871-4916 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
The first time is always free :) On Thu, Jul 30, 2009 at 1:50 PM, John Todd jt...@digium.com wrote: I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one. Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software. This is a time-expiring beta - the software will stop working on August 31. The download is also currently time-limited - it will be available until August 7 on our website. After the 31st, you would need to have purchased a license for the SfA software (sorry, no pricing that I can give you right now - that will be a separate announcement. I'm just the community guy - I have no idea about pricing or commercial contracts or the like, so please wait until that's been announced as I will find out about the same time as you do. :-) Trial purchase page: http://store.digium.com/productview.php?product_code=804-00019 JT --- John Todd email:jt...@digium.comemail%3ajt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Error
Hi All, I'm trying to test asterisk voicemail on recording my own unavailable message, busy message or temporary message. I was looking at the console and saw this message: app_voicemail store_file Memory map failed Then i looked at /var/spool/asterisk/ there were no recorded greetings. what does the error mean? TIA Regards Ron ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not getting inbound CallerID name on Asterisk
The following pastebin shows the inbound call, inbound INFO containing the Remote-Party-ID string, and the SIP acknowledgement of the INFO. Asterisk does not send the data from the Remote-Party-ID string on to the phone, nor does it set the CALLERID(name) variable after receiving the message. http://pastebin.com/m45e0adbd Thanks, Chris On Sun, Jul 26, 2009 at 1:19 PM, Chris Douglaschris.douglas at pioneerballoon.com wrote: We have an inbound PRI connected to our Cisco 3825 router which is then passing the calls to Asterisk as SIP calls. We're getting the CallerID number but not the CallerID name. We are seeing the name in the RPID field with a SIP trace on the Asterisk box but don't understand why it's not registering as the CallerID name. What do you get when you enable debugging on the asterisk cli? core set verbose 3 make a call. Do you see caller ID going through? What does your dialplan look like? You can use NoOp() calls to pop out values including a caller ID if it exists. -Original Message- From: Chris Douglas [mailto:chris.doug...@pioneerballoon.com] Sent: Sunday, July 26, 2009 12:20 PM To: 'asterisk-users@lists.digium.com' Subject: Not getting inbound CallerID name on Asterisk We have an inbound PRI connected to our Cisco 3825 router which is then passing the calls to Asterisk as SIP calls. We're getting the CallerID number but not the CallerID name. We are seeing the name in the RPID field with a SIP trace on the Asterisk box but don't understand why it's not registering as the CallerID name. Here is a link to pastebin with the Sip trace. In it you can see the RPID is seen from the Asterisk box but it is not used/sent to the phones. http://pastebin.com/m45e0adbd Here is the section from Sip.conf describing the Cisco 3825 connection. We have tried type as both friend and peer as it is now with no change. [cisco_3825] context=default type=peer host=10.0.0.10 disallow=all allow=g729 allow=ulaw allow=alaw trustrpid=yes sendrpid=no All phones are not receiving the CallerID name, here is a sample from sip.conf of a phone config. [8670] secret=8670 context=ict_sip type=friend host=dynamic call-limit=5 agentlogin=yes mailbox=8...@ictvm progressinband=no sendrpid=yes Any help is greatly appreciated! Thanks, Chris Douglas Technical Services Manager Pioneer Balloon Company ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] odd T1 issue
Howdy, Just installed a new switch in a new location (Ubuntu, 2.6.24-24 kernel, zaptel 1.4.12.1 built from source, libpri-1.4.10.1 built from source, asterisk 1.4.26 built from source, wanpipe 3.5.4 built from source, Sangoma A104d with firmware that is probably a year old). I plugged in an RBS T1, ESF, B8ZS, wink start, and MF signalling. I stuck with the defaults that the wanpipe build wrote in zaptel.conf when I told it ESF, B8ZS, and EM Wink: --- loadzone=us defaultzone=us #Sangoma A104 port 1 [slot:6 bus:2 span:1] wanpipe1 span=1,0,0,esf,b8zs em=1-24 --- But I changed the signalling line in zapata.conf because with signalling=em_w I couldn't place any outbound calls and all inbound calls, though they worked, were sending 33 as the DID number no matter what DID was called. So after trying a few different signalling methods, I found that: ;Sangoma A104 port 1 [slot:6 bus:2 span:1] wanpipe1 context=from-pstn group=0 ;signalling=em_w signalling=featb channel = 1-24 Allowed me to place outbound calls with no problems. Inbound calls, however, now do give me all the digits of the DID number dialed, but with the odd 33 interspersed. For example when the DID number is 715 7600 I get (in the CLI): Connected to Asterisk 1.4.26 currently running on vigw-crown1 (pid = 5515) Verbosity is at least 100 Core debug is at least 100 -- Starting simple switch on 'Zap/10-1' -- Executing [731573...@from-pstn:1] Dial(Zap/10-1, SIP/231) in new stack -- Called 231 -- SIP/231-081d1108 is ringing == Spawn extension (from-pstn, 731573600, 1) exited non-zero on 'Zap/10-1' -- Hungup 'Zap/10-1' And when the DID number is 715 7999, I get: -- Starting simple switch on 'Zap/11-1' -- Executing [731573...@from-pstn:1] Dial(Zap/11-1, SIP/231) in new stack -- Called 231 -- SIP/231-081d1108 is ringing == Spawn extension (from-pstn, 731573999, 1) exited non-zero on 'Zap/11-1' -- Hungup 'Zap/11-1' I matched all entries with _X. in extensions.conf and had them ring a SIP phone as you can see above, just to see what I was given by the carrier. It is perfectly consistent, so I *could* get by with matching the stupid 3's, but obviously I would rather not leave it this way. Any ideas? Thanks, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk Inter digit delay
On Thu, Jul 30, 2009 at 1:19 AM, hadi motamedimotamed...@gmail.com wrote: Thank you very much for your reply . But please be informed that our current line-outgoing route is being configured as the followings (in extensions.conf): Set(TIMEOUT(digit)=timeout) There's definitely more to your dialplan than the sample you provided. You need to add the timeout in there. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
On 7/30/09, Steve Totaro stot...@asteriskhelpdesk.com wrote: The first time is always free :) On Thu, Jul 30, 2009 at 1:50 PM, John Todd jt...@digium.com wrote: I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one. Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software. This is a time-expiring beta - the software will stop working on August 31. The download is also currently time-limited - it will be available until August 7 on our website. After the 31st, you would need to have purchased a license for the SfA software (sorry, no pricing that I can give you right now - that will be a separate announcement. I'm just the community guy - I have no idea about pricing or commercial contracts or the like, so please wait until that's been announced as I will find out about the same time as you do. :-) Trial purchase page: http://store.digium.com/productview.php?product_code=804-00019 JT --- John Todd email:jt...@digium.comemail%3ajt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
