[asterisk-users] start music on hold coredump
Hi the following is message,Any advice appreciated, thank you. (gdb) bt #0 0x00429410 in __kernel_vsyscall () #1 0x00bead80 in raise () from /lib/libc.so.6 #2 0x00bec691 in abort () from /lib/libc.so.6 #3 0x00c2324b in __libc_message () from /lib/libc.so.6 #4 0x00c2b883 in _int_malloc () from /lib/libc.so.6 #5 0x00c2d3ab in malloc () from /lib/libc.so.6 #6 0x00c21ff3 in vasprintf () from /lib/libc.so.6 #7 0x00c07efe in asprintf () from /lib/libc.so.6 #8 0x080a059f in build_filename (filename=0xb77b02d0 /var/lib/asterisk/mohmp3/wq//cn/wq5, ext=0xb77af1a0 WAV) at file.c:276 #9 0x080a18f5 in ast_filehelper (filename=0xb77b02d0 /var/lib/asterisk/mohmp3/wq//cn/wq5, arg2=0x0, fmt=0x0, action=ACTION_EXISTS) at file.c:445 #10 0x080a250b in fileexists_core (filename=0x9140ce0 /var/lib/asterisk/mohmp3/wq//wq5, fmt=0x0, preflang=0x2a7e Address 0x2a7e out of bounds, buf=0xb77b02d0 /var/lib/asterisk/mohmp3/wq//cn/wq5, buflen=38) at file.c:601 #11 0x080a28bd in ast_openstream_full (chan=0x9079408, filename=0x9140ce0 /var/lib/asterisk/mohmp3/wq//wq5, preflang=0x8fff593 cn, asis=1) at file.c:709 #12 0x001cb7bc in moh_files_generator (chan=0x9079408, data=0xb7c440b0, len=160, samples=160) at res_musiconhold.c:264 #13 0x080844a3 in ast_read_generator_actions (chan=0x9079408, f=0x8f8b0dc) at channel.c:1925 #14 0x08086b85 in __ast_read (chan=0x9079408, dropaudio=0) at channel.c:2315 #15 0x08073c99 in autoservice_run (ign=0x0) at autoservice.c:114 #16 0x0810072b in dummy_start (data=0x90e37f0) at utils.c:895 ---Type return to continue, or q return to quit--- #17 0x00d1349b in start_thread () from /lib/libpthread.so.0 #18 0x00c9342e in clone () from /lib/libc.so.6 -- Best regards! jordan pan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call sip:u...@domain.com?
On Thu, 16 Dec 2010 17:05:35 -0500, Jamie A. Stapleton jstaple...@computer-business.com wrote: Just add something like this to your dialplan: exten=1234,1,Dial(SIP/u...@domain.com) Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com. Thanks Jamie, but isn't there a universal way to solve this, so that users can dial any SIP number without first having to create an extension for that specific number? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pass DTMF to IVR gateway through SIP phone conferencing.
Hi friends, I want to implement following scenario using Asterisk. Please suggest me whether it is possible or not. This is bit off Asterisk and more on SIP side. An Asterisk box with one Station(SIP channel) and PRI. Agent dials a PSTN number of customer from station through Asterisk PRI. Agent gets connected with customer. Agent puts customer on hold. Agent dials another PSTN number which is of IVR gateway. Agent now makes conference(Station facility) with customer and IVR gateway. Gateway plays an IVR asking customer to enter his customer id number. My question is, will DTMF get forwarded to IVR gateway? I am asked to implement this and not having PRI for the moment in my Asterisk box. Thanking you in advance. -AsteriskMan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Tandberg VCS
Hi All, We have a Tandberg VCS System for Video conferencing and a customer running AsteriskNow (Asterisk 1.6 + FreePBX) for Audio conferencing. Problem Statement: How do we integrate the 2 systems such that Audio SIP calls are seamlessly passed between the two. Sorry we're just starting up so a bit of general advice, or a link to any document would be great! If anybody has done this - would appreciate any tips :) Thanks! Jake -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi show channels / how to get the call duration for active calls?
Hi, for dahdi-calls I can see the current calls with dahdi show channels. But where can I see the current call-duration or the call-start-time? dahdi show channel n does not show this info. -Thorsten- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call sip:u...@domain.com?
Le 17/12/2010 07:45, Gilles a écrit :
On Thu, 16 Dec 2010 17:05:35 -0500, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
Just add something like this to your dialplan:
exten=1234,1,Dial(SIP/u...@domain.com)
Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com.
Thanks Jamie, but isn't there a universal way to solve this, so that
users can dial any SIP number without first having to create an
extension for that specific number?
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
--
Daniel
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Re: [asterisk-users] How to find , internal, external inbound or outbound
reply please On 12/17/2010 10:03 AM, Nikhil wrote: Hi Does anyone knows how to find out a call in a asterisk is external incoming ,external out going or internal Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ported Asterisk in Android
Hi Does anyone ported Asterisk to Android OS .please give details Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Forwarding
Experiencing a problem when users attempt to forward a voicemail from within VoiceMailMain(Option 8) I see the console message: Couldn't not find mailbox XXX in context default As why are running in a multi-tenant environment voicemail.conf has been separated into individual contexts. The users retrieve their email by dialing an extension which calls VoiceMailMail(x...@vmcontext) so how do I instruct Asterisk to use that context when forwarding voicemails ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Contradiction between 2 AMI actions QueueSummary and Queuestatus
Guys, Why is such contradiction between 2 AMI actions QueueSummary and Queuestatus? Look at LoggedIn of QueueSummary and Event: QueueMember. Also LongestHoldTime of QueueSummary does not give correct value. - Action: QueueSummary Queue: retailBanking Response: Success Message: Queue summary will follow Event: QueueSummary Queue: retailBanking LoggedIn: 0 Available: 0 Callers: 0 HoldTime: 22 TalkTime: 231 LongestHoldTime: 0 Event: QueueSummaryComplete - Action: Queuestatus Queue: retailBanking Response: Success Message: Queue status will follow Event: QueueParams Queue: retailBanking Max: 0 Strategy: rrmemory Calls: 0 Holdtime: 22 TalkTime: 231 Completed: 5 Abandoned: 4 ServiceLevel: 0 ServicelevelPerf: 0.0 Weight: 0 Event: QueueMember Queue: retailBanking Name: agent2 Location: SIP/1110 Membership: dynamic Penalty: 0 CallsTaken: 1 LastCall: 1292570332 Status: 5 Paused: 0 Event: QueueMember Queue: retailBanking Name: agent1 Location: SIP/ Membership: dynamic Penalty: 0 CallsTaken: 3 LastCall: 1292581231 Status: 5 Paused: 0 Event: QueueStatusComplete - --AsteriskMan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HA: what is missing to keep ongoing calls during failover ?
Hi, What is currently missing in Asterisk ecosystem to get 2 servers active-active redundancy such as when server 1 is failing (in some circumstances), its ongoing calls (or most of them) are kept alive and handed over to server 2 ? I remember that a couple of years ago, Avaya claimed it could achieve this with its high end servers. Could it be possible with Asterisk ? Will SCF change this ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Contradiction between 2 AMI actions QueueSummary and Queuestatus
Asterisk Version: 1.8.0 Members are added through AddQueueMember in realtime Queues On Fri, Dec 17, 2010 at 4:52 PM, Asterisk Man theasterisk...@gmail.comwrote: Guys, Why is such contradiction between 2 AMI actions QueueSummary and Queuestatus? Look at LoggedIn of QueueSummary and Event: QueueMember. Also LongestHoldTime of QueueSummary does not give correct value. - Action: QueueSummary Queue: retailBanking Response: Success Message: Queue summary will follow Event: QueueSummary Queue: retailBanking LoggedIn: 0 Available: 0 Callers: 0 HoldTime: 22 TalkTime: 231 LongestHoldTime: 0 Event: QueueSummaryComplete - Action: Queuestatus Queue: retailBanking Response: Success Message: Queue status will follow Event: QueueParams Queue: retailBanking Max: 0 Strategy: rrmemory Calls: 0 Holdtime: 22 TalkTime: 231 Completed: 5 Abandoned: 4 ServiceLevel: 0 ServicelevelPerf: 0.0 Weight: 0 Event: QueueMember Queue: retailBanking Name: agent2 Location: SIP/1110 Membership: dynamic Penalty: 0 CallsTaken: 1 LastCall: 1292570332 Status: 5 Paused: 0 Event: QueueMember Queue: retailBanking Name: agent1 Location: SIP/ Membership: dynamic Penalty: 0 CallsTaken: 3 LastCall: 1292581231 Status: 5 Paused: 0 Event: QueueStatusComplete - --AsteriskMan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call sip:u...@domain.com?
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
Thanks Daniel. I added that line above, told Asterisk to reload the
dialplan, and typed the following in XLite:
9*031...@ekiga.net
This is to perform an echo test
http://wiki.ekiga.org/index.php/Fun_Numbers
I guess something else must be done to Asterisk for this to work:
==
CLI
-- Executing [9*031...@my-phones:1] Dial(SIP/6011-00a1b67c,
SIP/*031600) in new stack
[Dec 17 11:43:14] WARNING[306]: chan_sip.c:2923 create_addr: No such
host: *031600
[Dec 17 11:43:14] WARNING[306]: app_dial.c:1183 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/6011-00a1b67c' status is
'CHANUNAVAIL'
==
I also tried this, same result:
exten=_9.,1,Dial(SIP/ippi_outgoing/${EXTEN:1})
Do I need to add something in sip.conf, or some other configuration
file?
Thank you.
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Re: [asterisk-users] dahdi show channels / how to get the call duration for active calls?
You probably want "core show channels verbose".Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brasil+55 54 2104-7000Information Security ManagerCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brazil+55 54 2104-7000Hi,for dahdi-calls I can see the current calls with "dahdi show channels". But where can I see the current call-duration or the call-start-time? "dahdi show channel n" does not show this info.-Thorsten---_-- Bandwidth and Colocation Provided by http://www.api-digital.com --New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/helloasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find , internal, external inbound or outbound
- Original Message - reply please On 12/17/2010 10:03 AM, Nikhil wrote: Hi Does anyone knows how to find out a call in a asterisk is external incoming ,external out going or internal Thanks Nikhil Perhaps if you were clearer in the question you are asking ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call sip:u...@domain.com?
On Thu, 16 Dec 2010 11:54:31 +0100, Gilles codecompl...@free.fr wrote: Now, I'd like to be able to call any number on the Net that is advertised as sip:u...@domain.com, such as those: I mean: Do I really have to first create a section in sip.conf each time a user needs to call a number on a new SIP server? http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find , internal, external inbound or outbound
There's no such concept in Asterisk. Everything is a call, doesn't matter its direction.Atenciosamente,Vinícius FontesGerente de Segurança da InformaçãoCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brasil+55 54 2104-7000Information Security ManagerCanall Tecnologia em ComunicaçõesPasso Fundo - RS - Brazil+55 54 2104-7000reply pleaseOn 12/17/2010 10:03 AM, Nikhil wrote: Hi Does anyone knows how to find out a call in a asterisk is external incoming ,external out going or internal Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--_-- Bandwidth and Colocation Provided by http://www.api-digital.com --New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/helloasterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call sip:u...@domain.com?
Le 17/12/2010 12:48, Gilles a écrit :
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
Thanks Daniel. I added that line above, told Asterisk to reload the
dialplan, and typed the following in XLite:
9*031...@ekiga.net
This is to perform an echo test
http://wiki.ekiga.org/index.php/Fun_Numbers
I guess something else must be done to Asterisk for this to work:
==
CLI
-- Executing [9*031...@my-phones:1] Dial(SIP/6011-00a1b67c,
SIP/*031600) in new stack
[Dec 17 11:43:14] WARNING[306]: chan_sip.c:2923 create_addr: No such
host: *031600
[...]
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
You should read info on voip.org to learn basis of Asterisk.
--
Daniel
--
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Re: [asterisk-users] Call sip:u...@domain.com?
On 10-12-17 06:48 AM, Gilles wrote:
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
Thanks Daniel. I added that line above, told Asterisk to reload the
dialplan, and typed the following in XLite:
9*031...@ekiga.net
This is to perform an echo test
http://wiki.ekiga.org/index.php/Fun_Numbers
You have to tell it the host to request the extension from. All you're doing is
dialing SIP/*031600, which with that format, is going to try and call [*031600]
as defined in sip.conf.
You're missing the host that you want to call. The format needs to be
SIP/*031600@some_hostname
What you're trying to do is essentially what FreeNum was designed for:
http://www.freenum.org
We discuss it in this chapter here:
http://ofps.oreilly.com/titles/9780596517342/ch12.html
Leif.
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Re: [asterisk-users] How to find , internal, external inbound or outbound
I HATE OUTLOOK _ reply please On 12/17/2010 10:03 AM, Nikhil wrote: Hi Does anyone knows how to find out a call in a asterisk is external incoming ,external out going or internal Thanks Nikhil _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vinícius Fontes Sent: Friday, December 17, 2010 6:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to find , internal, external inbound or outbound There's no such concept in Asterisk. Everything is a call, doesn't matter its direction. Rant at the top, heres my actual thoughts. Yes, Everything is a call, BUT each call has an origin and at least one destination. If I do this: Exten = 1234,1,answer Exten = 1234,n,playback(tt-weasels) Exten = 1234,n,hangup This call doesnt actually go anywhere, but it has an origin of the phone I dial 1234 from and a destination of a local channel to process the playback. If I add to my incoming context Exten = s,n,goto(default,1234,1) And call my Asterisk using a SIP or DAHDI trunk, the origin is that trunk, and destination is still a local channel. Now If I actually speak to someone, the destination will be the SIP or DAHDI channel that asterisk reaches them on. If you look in the CDR, the origin and destination fields will tell you which trunk each uses. By definition (mine, but its probably somebody elses too), this is my answer to OPs question Origin destination = Sip trunk SIP ext external incoming DAHDI trunk SIP ext external incoming SIP extSIP trunk external out SIP extDAHDI trunk external out SIP extSIP extinternal Local channel = SIP ext There are smarter people than me that read this list; perhaps they will improve on this answer. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find , internal, external inbound or outbound
Hi, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack http://pastebin.com/tbjh5qzP regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attack problem
HI, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack http://pastebin.com/tbjh5qzP regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Freeze In 1.4 realtime
Has anyone seen the following in 1.4 (1.4.17) We have istances when the number of sip channels in use multiples up (eg: we have 40 channels in use, and then it will jump to 80, then 100+ and it will keep going upwards) and in doing this, all the channels which are in use at that time are simply cut off or frozen. The only way for us to get everything back to normal is via a hard restart of asterisk. If you have seen this behaviour or know of a bug which is associated to this can you advise, or point us at a fix. Thanks -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA: what is missing to keep ongoing calls during failover ?
On 10-12-17 06:17 AM, Olivier wrote: Hi, What is currently missing in Asterisk ecosystem to get 2 servers active-active redundancy such as when server 1 is failing (in some circumstances), its ongoing calls (or most of them) are kept alive and handed over to server 2 ? I remember that a couple of years ago, Avaya claimed it could achieve this with its high end servers. Could it be possible with Asterisk ? Will SCF change this ? That's exactly what the demo at AstriCon showed SCF could do. Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wireless Desktop VoIP Phone?
I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target
The setup is this: 2 sip handsets (a Cisco 7960 and a 7961) exten 401/402 1 fxs/dahdi cordless phone, exten 201 rhino fxo/fxs analog card asterisk 1.4.31 This is running on a Soekris 5501 with Astlinux 0.7.2 While I do have FXO capabilities, no POTS lines are connected. When a call comes in (VoIP, either SIP or IAX) it is usually answered on one of the SIP Cisco phones(x 401 or 402). If it is for my wife, then I would like to transfer the call to the fxs/dahdi analog cordless phone (x 201). At one time this worked, but about a year or so ago it stopped. What is happening now is that the call comes in (x 401), is transferred via the cisco transfer soft button to (x 201), ... during this time the caller was put on hold or rather was automatically connected to the MOH process... , When (x 201) answers the phone, they are connected to the MOH process and cannot hear or talk to the original caller. In testing, if I leave the (x 201) call open, the original outside call is kept open as well (the original caller hears nothing). A look at the active sessions confirms this. When either (x 201) or original caller hang up, the call/connection is terminated. I can transfer calls from one Cisco to the other without issue. I have looked around at my configs, but don't see anything that would cause this... but truthfully I don't even know where to begin with something like this. I checked the logs to see if there was something helpful there but did not see anything. My only though is that it is something with the way the Cisco internal transfer process happens... but again, I don't know where to begin to test that theory. Has anyone seen or heard of this? Know how to resolve to expected behavior? I appreciate any pointers. John R. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call sip:u...@domain.com?
On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
Thanks for the tip but I wanted to be able to call _any_ SIP number,
not just Ekiga, so needed a destination-agnostic solution.
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Re: [asterisk-users] setting up callerid
Hi Dave,
On Thu, Dec 16, 2010 at 1:52 PM, dave george dgeo...@teletoneinc.com wrote:
Tried the following but no luck:
exten = _53.,1,Set(CALLERID(num)=473520)
exten = _53.,n,Dial(SIP/${ext...@ss74)
I am still passing IMSI310410381554227 as the CALLERID.
My peer is setup as follows:
[IMSI310410381554227]
canreinvite=no
type=peer
context=openbts
callerid=473520
I see you are using OpenBTS. To my understanding, OpenBTS does not
support caller ID, so I don't think it can work.
But as I have the same issue as you, I'd be glad to be wrong ! :D Let me know.
Regards
Axelle
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Re: [asterisk-users] Call sip:u...@domain.com?
On Fri, 17 Dec 2010 09:00:39 -0500, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
You have to tell it the host to request the extension from. All you're doing
is
dialing SIP/*031600, which with that format, is going to try and call
[*031600]
as defined in sip.conf.
You're missing the host that you want to call. The format needs to be
SIP/*031600@some_hostname
Thanks for the tip. Elsewhere, someone suggested adding this code in
extensions.conf, which solved the problem:
===
[macro-dialsipuri]
exten = s,1,Set(dialuri=${CUT(ARG1,\;,1)})
exten = s,n,Verbose(Calling SIP URI ${dailuri})
exten = s,n,Verbose(--- From: ${CALLERID(all)})
exten = s,n,Dial(SIP/${dialuri},60,tr)
exten = s,n,Congestion()
[internal]
...
exten = _[a-z].,1,Macro(dialsipuri,${ext...@${sipdomain})
exten = _[A-Z].,1,Macro(dialsipuri,${ext...@${sipdomain})
===
What you're trying to do is essentially what FreeNum was designed for:
http://www.freenum.org
I'll read up on Freenum, but I was just trying to do something that I
thought was very simple, namely make a phone call over the Net, ie.
have XLite send an INVITE to Asterisk, which would then forward the
INVITE to the remote server, which would ring the phone. I expected
Asterisk users to make direct calls routinely, but maybe it's not that
frequent.
We discuss it in this chapter here:
http://ofps.oreilly.com/titles/9780596517342/ch12.html
Thanks Leif. I was going through the 2nd edition, which doesn't seem
to deal with direct, Internet dialing. I'll go through that Chapter 12
in the 3rd edition.
Thank you.
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Re: [asterisk-users] Attack problem
- Original Message - HI, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack http://pastebin.com/tbjh5qzP regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ask on an IRCD list ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attack problem
On Friday 17 Dec 2010, Khaled W. Chehab wrote: HI, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack # /etc/init.d/ircd stop # chmod -x /etc/init.d/ircd Should do the business :) -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Freeze In 1.4 realtime
Ishfaq Malik wrote: Has anyone seen the following in 1.4 (1.4.17) 1.4.17 is quite old, I'd suggest running the most current 1.4.38 and see if it fixes your problem. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to block everyone outside of our lan
I'd like to find out how to block everyone outside of the our LAN. The following phone call got through: 1. accountcode: Blank 2. src: Caller*ID number Blank 3. dst: Destination extension 901185294464086 4. dcontext: Destination context DLPN_DialPlan1 5. clid: Caller*ID with text Blank 6. channel: Channel used SIP/xxx-088c48d8 7. dstchannel: Destination channel DAHDI/1-1 8. lastapp: Last application if appropriate Dial 9. lastdata: Last application data (arguments) Dahdi/g1/01185294464086 10. start: Start of call 2010-12-16 04:49:28 11. answer: Answer of call 2010-12-16 04:49:32 12. end: End of call 2010-12-16 04:49:52 13. duration: Total time in system, 24seconds 14. billsec: Total time call is up, 20seconds 15. disposition: What happened to the call: ANSWERED 16. amaflags: What flags to use: DOCUMENTATION In Sip.conf I have: deny=0.0.0.0/0.0.0.0 permit=192.168.1.201/255.255.255.255 All the other local phones here snip One WanIP address Thank you, Gary Kuznitz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Attack problem
On Fri, 17 Dec 2010, Khaled W. Chehab wrote: How can I stop the ircd attack This isn't an Asterisk issue. 0) Turn off your IRC service. 1) Add some rules to iptables. 2) Investigate fail2ban and see if it is an appropriate response. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Freeze In 1.4 realtime
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Friday, December 17, 2010 9:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Freeze In 1.4 realtime Has anyone seen the following in 1.4 (1.4.17) We have istances when the number of sip channels in use multiples up (eg: we have 40 channels in use, and then it will jump to 80, then 100+ and it will keep going upwards) and in doing this, all the channels which are in use at that time are simply cut off or frozen. The only way for us to get everything back to normal is via a hard restart of asterisk. If you have seen this behaviour or know of a bug which is associated to this can you advise, or point us at a fix. Thanks Ishfaq Malik Sip reload doesn't make this go away? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
On Fri, Dec 17, 2010 at 12:40 PM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- Many phones like the snom 870 include a USB connector for a wireless adapter. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call sip:u...@domain.com?
On Fri, Dec 17, 2010 at 04:52:32PM +0100, Gilles wrote: Thanks for the tip but I wanted to be able to call _any_ SIP number, not just Ekiga, so needed a destination-agnostic solution. How would you _expect_ to be able to specify a destination server from a telephone keypad? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
On Fri, 17 Dec 2010, Matt wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? DECT? (as in Siemens Gigaset) Or are you looking for a box with handset that you can lift and a dialpad/display on the base type of thing? Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
At 07:40 AM 12/17/2010, you wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? Possibly one of the Aastra phones, 480i-CT or maybe a 57i-CT. Ira -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
Nortel 1535. Does video as well. On Fri, Dec 17, 2010 at 10:40 AM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call sip:u...@domain.com?
Le 17/12/2010 16:52, Gilles a écrit :
On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
Thanks for the tip but I wanted to be able to call _any_ SIP number,
not just Ekiga, so needed a destination-agnostic solution.
You can use SipBroker.
http://www.sipbroker.com/sipbroker/action/providerWhitePages
--
Daniel
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Re: [asterisk-users] Voicemail Forwarding
Is that user trying to forward to xxx in the same context? On Fri, Dec 17, 2010 at 5:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: Experiencing a problem when users attempt to forward a voicemail from within VoiceMailMain(Option 8) I see the console message: Couldn't not find mailbox XXX in context default As why are running in a multi-tenant environment voicemail.conf has been separated into individual contexts. The users retrieve their email by dialing an extension which calls VoiceMailMail(x...@vmcontext) so how do I instruct Asterisk to use that context when forwarding voicemails ? -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find , internal, external inbound or outbound
This list is being attacked by some Khaled I guess. How can we stop him? On Fri, Dec 17, 2010 at 9:34 AM, Khaled W. Chehab kche...@xplorium.com wrote: Hi, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack http://pastebin.com/tbjh5qzP regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
Snom Sent from Droid On Dec 17, 2010 12:36 PM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
On Fri, 17 Dec 2010 10:40:00 -0500, Matt wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? I beleive that snom supports the use of a wifi usb dongle in the 8x0 series phones. Also, Linksys/Cisco offered an 802.11g adapter that could be paired with their phones, making them wifi capable. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
On Fri, 2010-12-17 at 10:40 -0500, Matt wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? Linksys Cisco SPA525 has integrated WiFi and Bluetooth Snom 820 or 870 with optional USB adapter -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
Cisco also make a wireless adapter for the 500 series phones. On Fri, Dec 17, 2010 at 7:40 AM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to block everyone outside of our lan
On Fri, 17 Dec 2010, Gary Kuznitz wrote: I'd like to find out how to block everyone outside of the our LAN. The following phone call got through: 0) What makes you think it came from outside? 1) iptables/fail2ban 2) bind Asterisk to the IP address of the 'inside' interface. Search for 'bind' in sip.conf. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
I use grandstream with the linksys/cisco adapter. Bryant On Dec 17, 2010, at 3:04 PM, Michael Graves mgra...@mstvp.com wrote: On Fri, 17 Dec 2010 10:40:00 -0500, Matt wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? I beleive that snom supports the use of a wifi usb dongle in the 8x0 series phones. Also, Linksys/Cisco offered an 802.11g adapter that could be paired with their phones, making them wifi capable. Michael -- Michael Graves mgravesatmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo Cancellation Problem - Invalid Argument?!?
Trying again... I think this got lost in the mailing list interruptions during the last day or two... - Forwarded Message - From: Tim Nelson tnel...@fudnet.net To: asterisk-users@lists.digium.com Sent: Wednesday, December 15, 2010 5:07:20 PM Subject: [asterisk-users] Echo Cancellation Problem - Invalid Argument?!? Greetings folks- I'm experiencing issues with a freshly installed box. When a call comes in via PRI (Sangoma AFT-A104), I see this in my logs: [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 12 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 8 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 10 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 9 (Invalid argument) Relevant components: Asterisk: Asterisk SVN-trunk-r290509 built by root @ prigw01 on a i686 running Linux on 2010-11-30 22:12:05 UTC DAHDI: dahdi-linux-complete-2.4.0+2.4.0 LibPRI: libpri-1.4.11.5 Wanpipe: wanpipe-3.5.18 Kernel: Linux prigw01 2.6.32-24-generic #39-Ubuntu SMP Wed Jul 28 06:07:29 UTC 2010 i686 GNU/Linux The card does not have a hardware echo canceler. It should use MG2 as specified in DAHDI's system.conf: #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit #autogenrated on 2010-12-08 #Dahdi Channels Configurations #For detailed Dahdi options, view /etc/dahdi/system.conf.bak loadzone=us defaultzone=us #Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 span=1,1,0,esf,b8zs bchan=1-23 #dchan=24 echocanceller=mg2,1-23 hardhdlc=24 And, from chan_dahdi.conf: ;Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 switchtype=national context=ldrouted group=1 echocancel=yes signalling=pri_net channel =1-23 Any thoughts, pointers, suggestions? The echo is horrible, please help me make it stop. :-) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
Grandstream GXV3140 has a WiFi USB adapter. Alex Saavedra On Fri, Dec 17, 2010 at 11:40 AM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless Desktop VoIP Phone?
I belive the WBP54g cisco/LINKSYS adapter is what we are using with the Grandstream phones. You have to buy a Cisco/Linksys power supply but it works great. I have over 200 of them out there. Bryant From: Jeremy Betts jer...@freevoicepbx.com Sent: Friday, December 17, 2010 4:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Wireless Desktop VoIP Phone? Cisco also make a wireless adapter for the 500 series phones. On Fri, Dec 17, 2010 at 7:40 AM, Matt mhop...@gmail.com wrote: I'm looking for a wireless desktop VoIP phone. Does any exist? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PTMP BRI Unable to receive TEI from network in state 2(Assign awaiting TEI) - Asterisk 1.6.2, Latest DAHDI, LibPRI
On 17/12/10 5:56 PM, Olivier wrote: Hi, Did you use libpri 1.4.11.5 or 1.4.12-beta ? Recently l tried 1.4.11.5 on a live system and it failed (Asterisk 1.6.1.18 and dahdi trunk, Junghanns QuadBRI, PtmP lines). Going back to 1.4.11.2 solved it. Unfortunately, I couldn't note what error message were then generated. Heh, latest everything - so LibPRI trunk. I did try going backwards in terms of DAHDI, but not LibPRI - will try that on Monday. By the way, Kevin/Russell etc, any chance we could get a test added to bamboo for physical connectivity? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
