Displaying 20 results from an estimated 1000 matches similar to: "dreaded one-way audio with nat=yes"
2012 Aug 02
4
html/js/flash/air SIP clients?
Dear list,
I am looking for an open source SIP client(or any SDK) that can work on a
browser. It may be based html5, javascript, flash, adobe air. I have done
some research myself and I would like to ask the community if they have any
further hints for me. Real life experience would be awesome.
Thanks,
Regards,
Arstan Jusupov
-------------- next part --------------
An HTML attachment was
2010 Oct 23
3
Why such high latency on internal lan?
My internal lan is small, 100mb, all wired. aastra phones.
sip show peers
.......
142/... 10.10.10.42 D A 5060 OK (136 ms)
144/... 10.10.10.44 D A 5060 OK (138 ms)
145/... 10.10.10.45 D A 5060 OK (133 ms)
But pings are < 1ms:
ping 10.10.10.42
........
rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms
Why are the sip latencies so
2010 Oct 21
1
Why high latency on internal lan?
I have a 100MB internal lan. aastra's are wired. asterisk box is wired
next to the switch. But look:
sip show peers
........
142/142 10.10.10.42 D A 5060 OK (137 ms)
144/144 10.10.10.44 D A 5060 OK (136 ms)
145/145 10.10.10.45 D A 5060 OK (168 ms)
150/150 10.10.10.50
2010 Nov 10
2
Asterisk 1.8 -- queue not recognizing that agent is busy
Hi All,
I've got a realtime queue in place (strategy is "wrandom"), and have
added a member dynamically via "queue add member ". My agent shows in
the queue, but when he gets the call is not recognized as "In Use".
Here is the output from "queue show" prior to the call:
*CLI> queue show
QUEUE_3 has 0 calls (max unlimited) in 'wrandom'
2008 Dec 18
1
[Fwd: Asterisk client for ekiga.net NAT problem]
I am experiencing a "606 not Acceptable" error trying to set up an
Asterisk server as an ekiga.net client. My server is behind a firewall
with NAT routing. I have googled this problem and read about Asterisk
feeding its local ip address to ekiga.net. That seems to be my
problem.
I tried putting stunaddr=stun.ekiga.net into the sip.conf file under
[ekiga]. I also tried
2005 Apr 22
5
IAX help
I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2
channel. However the call is being rejected on the (telx-nyc) server.
See error below copied from telx-nyc CLI>
Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read:
Rejected
connect attempt from 192.168.0.251
I have icluded the following conf files
1. extensions.conf (telx-nyc)
2. iax.conf (telx-nyc)
3.
2007 Oct 23
2
register => to let Asterisk register to another softswitch via SIP
Dear Alex;
Thanks alot for your nice help.
This is if I need to let Asterisk register with
another softswitch (so I used register =>), what if I
need asterisk to send call for the softswitch without
register to it (directly)? If I removed the register
=> then how it will distiguish the IP address in the
"host" at the [sip_trunk] is the IP address of the
softswitch that need to
2005 Jul 12
3
Unable to call certain 800 numbers through Teliax
We are unable to call certain 800 numbers through Teliax but I thought I
would post this here and see if anyone else had the same problem with either
Teliax or other carriers.
The 800 numbers causing problems pick-up the call right away and are in the
US - American Airlines (8004337300) and Staples (800-378-2753) - we can call
many other 800 numbers just fine.
Our asterisk setup has a 4-port
2006 Mar 30
9
How is Teliax ?
Hi
I am looking at purchasing some DID lines from Teliax to install it on my
asterisk.
i would like to know some feed back on "Teliax" before i purchase.
suggest me if there are better sevice providers.
thanks
Giridhar Bandi
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2005 Oct 17
2
Teliax IAX problems -- Asterisk doesn't see answer
Not to point the finger at Teliax, but I'm having some unique problems with
their service that are as yet unexplained.
Incoming calls are fine.
Outgoing calls don't work, though they did at one time. As of today, I'm
running the latest code from CVS.
-- Called teliax/13143212222
-- Call accepted by 208.139.204.245 <http://208.139.204.245> (format ulaw)
-- Format for call is
2006 Jan 31
7
Teliax - Codec Preference effective?
Has anyone had problems getting their preffered codecs on the Teliax web
interface taking effect?
I have two accounts, two separate yet similarly configured * servers. On one
account the settings took right away - on another server I am getting no
result. In fact, no matter what I change the settings to, only the old
codecs are usable (otherwise * says it can't negotiate a codec). Teliax
2009 Nov 27
1
Asterisk 1.6.2.0-rc6 + Teliax = First Part Of Audio File Playback Cut Off
Good evening all, hope everyone in the US had a nice Thanksgiving!
On one of our internal servers, I decided to make the leap from 1.4.2x
to 1.6.2.0-rc6 so I could start learning about the changes and new
features that have been implemented. I upgraded all the configs, removed
all the deprecated stuff, etc -- well went well.
However, I noticed after the upgrade, when dialing into an
2005 Jun 29
2
Recommend against Teliax as primary ITSP
I really hate to have to make a post like this, but I feel I have little
choice but to relay to the group my experience with Teliax, and explain why
I recommend against using them as a primary Voip-> PSTN provider. I hope
that a letter like this will inspire companies like Teliax to work harder at
customer service, as well as circuit stability. We need more companies that
offer the types of
2005 Jun 17
1
Unable to find a path from g729 to gsm
Greetings! to all
Now, with some hard time and help from many genurous people's in the list, I
have come to this point with my TDM20B card & my teliax's IAX2 account.
I hope someone may help me with this issue mentioned below. I have already
selected my codec as gms in my iax.conf as well as in teliax's "my account
page" but still i have the same error when I attempt
2005 Oct 07
2
Teliax users, g729 question
I am using Teliax to terminate my calls, and I have 3 licenses' for
g729 from Digium. "show translations" verifies that the registration
took place.
When I place a call, having "allow=g729" as the only allow option in
iax.conf, I get the following error:
WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by
208.139.204.228: Unable to negotiate codec
If I place a
2005 Jun 08
2
Incoming call stops at random with Teliax
We are setting up asterisk with Teliax and having trouble getting the
incoming call to work all the time, the outgoing does not seem to have a
problem.
I have worked with their support but since they say that we are getting the
initial call to our server they want to charge to take a look.
They did a tcpdump and we are seeing an attempt but no CLI most of the time.
Some times we see this but it
2005 Jun 08
3
More than one account from the same provider?
I have had good success with my efforts to send faxes over voip using ulaw,
surprisingly, and I want to move it from testing to reality. I have an
account with Teliax, who have been very good. For voice I use g729 and ulaw,
but for faxing I can only allow ulaw. However, Teliax only sets the codec
preferences by account. I have another account, but I can't see a way to
register two accounts
2005 Feb 10
2
TelIAX troubles
We are having issues setting up our Asterisks server with Teliax
service. We are able to place calls, but cannot seem to get our
Asterisk box to answer from Teliax service.
We are using Asterisk with the latest AMP interface.
Teliax's examples are for single SIP phone, not the voice response
systems, nor do they provide any support of Asterisk other than basic
sample scripts...
2007 Aug 02
6
Teliax Quality of Service
Asterisk Users,
I recently ran into some problems with the quality of service with Teliax.
This occurred on August 1, 2007 with a dropped outbound call, audio
quality isse on the callee side- not hearing me well on callee side, and
sending DTMF tones (configured for RFC2833). Am I the only Teliax customer
having this problem?
It seems like when I am ready to go live with my Asterisk
2005 Jun 29
5
Problems with OR Logic in the GotoIf Statement
Skipped content of type multipart/alternative-------------- next part --------------
A non-text attachment was scrubbed...
Name: smime.p7s
Type: application/x-pkcs7-signature
Size: 3034 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050629/596126bc/smime.bin