When someone helps you, at least let them know if the problem is resolved or not

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I'll keep this brief because I don't want to come across like any more of an
a$$ than I absolutely have to, especially since I know I've blown my stack
before..... Gentlemen (and Ladies, if you're out there), If someone gives you advice on this list, and ESPECIALLY if they give you
advice offlist, have the courtesy to (AT THE LEAST) to let them know when/if
you get your question answered or your problem solved. As many people point out, on community supported mailing lists and forums
around the world, these user lists are comprised…

Asterisk Users 4.4 years ago 22 Answers

iax2 Max retries exceeded to host

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We have IAX2 peer between two asterisk and I am getting following error following IAX2 WARNING. IAX calling is functional [May 10 15:23:34] WARNING[2056]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3030332, seqno=211)
[May 10 15:23:44] WARNING[2047]: chan_iax2.c:3487 __attempt_transmit: Max retries exceeded to host 172.24.146.51 on IAX2/orasebcam-612 (type = 6, subclass = 11, ts=3040332, seqno=212)
[May 10 15:23:45] NOTICE[2058]: chan_sip.c:23826 handle_request_register: Registration from '' failed for '172.30.245.85:5060' - No matching peer found
[May 10 15:23:45] NOTICE[2058]: chan_sip.c:23826 handle_request_register: Registration from '
' failed for '172.30.245.85:5060'…

Asterisk Users 4.4 years ago 0 Answers

About X100P and TDM400P analog card in China

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So does this mean no solution when used ZAP/DAHDI with PSTN line? If I installed an E1, will that work?
Thanks.
Regards. On Wed, May 11, 2011 at 12:57 AM, John Novack > wrote: > Remember that ZAP/DAHDI channels don't receive ( because most PSTN/POTS
> lines don't provide ) answer supervision.
> This will certainly complicate what you want do do.
>
> John Novack
>
>
> Scott Zhang wrote:
>
> Hello. All.
> I am a bit new to asterisk,…

Asterisk Users 4.4 years ago 2 Answers

Using MixMonitor()

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Hello Folks; I appreciate all of the help so far - thanks. Another question: I am using MixMonitor() to record calls and I would
like to include the called number/extension in the filename: In my dialplan, I am able to save the file with the caller id in the
filename. However, what I am a little unsure about is the incoming
number/called number/extension - passing that information on to part of
the filename. Does anyone follow me? Glen

Asterisk Users 4.4 years ago 1 Answer

Asterisk 1.8.4 Now Available

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The Asterisk Development Team has announced the release of Asterisk 1.8.4. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.4 resolves several issues reported by the community.
Without your help this release would not have been possible. Thank you! Below is a sample of the issues resolved in this release: * Use SSLv23_client_method instead of old SSLv2 only.
(Closes issue #19095, #19138. Reported, patched by tzafrir. Tested by russell
and chazzam. * Resolve crash in ast_mutex_init()
(Patched by twilson) * Resolution of several DTMF based attended transfer…

Asterisk Users 4.4 years ago 4 Answers

Jabber / GTalk / hints

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On 04/17/2011 02:28 AM, Stefan Gofferje wrote:
> Hi!
>
> Are hints not yet implemented in res_jabber?
> I have this here:
>
> exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx@gmail.com
>
> But the hint doesn't show any difference. It always shows online on the
> phone and core show hints always shows that:
>
> 6003@internal : SCCP/6003 State:Unavailable Watchers 0
> 6002@internal : SCCP/6002 State:Idle Watchers 0
> 6001@internal : SCCP/6001 State:Idle Watchers 0
> 6000@internal : SCCP/6000 State:Idle Watchers 0
> 6004@internal…

Asterisk Users 4.4 years ago 0 Answers

40sec between dial execution and sending SIP request

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thanks,
this delay is occurred on asterisk server, between dial execution and
"CALLED ....."
On Mon, May 9, 2011 at 7:12 PM, Warren Selby wrote: > On Mon, May 9, 2011 at 7:26 AM, Pezhman Lali wrote:
>
>> Dear
>> I have a small pbx with asterisk 1.6.2.16.
>> I have a funny problem, there is exactly 40sec between dial execution and
>> sending first invite packet on sip.
>> do you have any idea where the problem is ?
>>
>

Asterisk Users 4.4 years ago 0 Answers