Hello,
I am using a Swiss VoIP provider called sipcall. They have what they
call a SIP trunk, and it is less expensive than individual accounts. From
Asterisk's point of view, this is just a regular SIP account, which
can however receive and send calls from multiple numbers. I just migrated
from i
On Wed, Sep 4, 2019, at 6:01 AM, bilal ghayyad wrote:
> Hello;
>
> I am facing a trouble with the SIP service provider, they are saying
> that there is a problem related to message option 200 (the heartbeat),
> so what is required to add this for the sip configuration? Below is my
> sip debug t
Hello;
I am facing a trouble with the SIP service provider, they are saying that there
is a problem related to message option 200 (the heartbeat), so what is required
to add this for the sip configuration? Below is my sip debug trace log with the
them and the sip peer configuration:
[Sep 4 12
On Tue, Jul 23, 2019, at 2:53 PM, Jerry Geis wrote:
> > rtp set debug on" will show the RTP traffic flowing,I did not see anything
> > printed when I pressed a key. I say the audio prints.
That means either it was not negotiated or was not picked up by Asterisk.
--
Joshua C. Colp
Digium - A Sa
> rtp set debug on" will show the RTP traffic flowing,
I did not see anything printed when I pressed a key. I say the audio prints.
Jerry
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Check out th
On Tue, Jul 23, 2019, at 1:47 PM, Jerry Geis wrote:
> I have a sip trunk between two asterisk boxes.
> I can call into the first box, hit 499 for example and the call goes to
> the second box and answers as expected plays me audio message just fine
> etc... My issue is that DTMF does not seem to
I have a sip trunk between two asterisk boxes.
I can call into the first box, hit 499 for example and the call goes to the
second box and answers as expected plays me audio message just fine etc...
My issue is that DTMF does not seem to be working.
Both sides are set for:
dtmfmode=RFC2833
What mi
On Thursday 04 January 2018 at 01:27:59, bilal ghayyad wrote:
> Hello
> It will be amazing if possible to do sip trunk with any of social media
> providers like: whatsapp, facebook, imo, viber, ... etc
To the best of my knowledge none of the services you mention either operate
over SIP or provid
Hello
It will be amazing if possible to do sip trunk with any of social media
providers like: whatsapp, facebook, imo, viber, ... etc.Did anyone has luck
with this? RegardsBilal
Sent from Yahoo Mail on Android--
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> Not sure maybe there's a better solution but I thought about using another
> peer with type=user for incoming connections.
That's what I've done for my connection to the service provider
I use (Vitelity), as they have different inbound and outbound
hosts/proxies. This works fine.
--
_
Not sure maybe there's a better solution but I thought about using another
peer with type=user for incoming connections.
On Mon, May 22, 2017, 6:13 PM Benoit Panizzon
wrote:
> Hello List
>
> I work at an SIP Provider and we have added and SBC in front of our
> Voice Switch to protect it.
>
> Thi
Hello List
I work at an SIP Provider and we have added and SBC in front of our
Voice Switch to protect it.
This requires all our SIP Trunk customers to register via a 'proxy'.
I struggle with Asterisk to work over a proxy.
This is what I have done so far.
register => usern...@sip.example.com:p
Sounds like a firewall setting to me. If you can ping, then Internet
Control Message Protocol (ICMP) packets are allowed, but if SIP traffic is
returning the ICMP Type 3 (code 13) response, then your SIP ports are
blocked (at least the firewall admin was nice enough to leave the reason
code messag
Hello,
I've got a remote system that is plagued with a strange issue.
It happens from time to time.
Yet, I've not found any condition that trigger this phenomenon.
Here is my setup:
- PSTN <---> ITSP <--SIP trunk--> Router <> Switch <> Asterisk
box
|
|
On Tuesday 26 Jul 2016, Jerry Geis wrote:
> It seems I am not getting any digits coming over a SIP trunk.
>
> How can I match "anything" or "nothing" and start my extension.
>
> Usually I have something like:
> exten => 55,1,Goto(,yyy,1)
>
> but if 55 does not come across and it appears to b
Hi Jerry,
In article ,
Jerry Geis wrote:
>
> It seems I am not getting any digits coming over a SIP trunk.
>
> How can I match "anything" or "nothing" and start my extension.
>
> Usually I have something like:
> exten => 55,1,Goto(,yyy,1)
>
> but if 55 does not come across and it appears
_. ?
Regards,
Marcelo H. Terres
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres
On Tue, Jul 26, 2016 at 11:39 AM, Jerry Geis wrote:
> It seems I am not getting any digits coming over a SIP trunk.
It seems I am not getting any digits coming over a SIP trunk.
How can I match "anything" or "nothing" and start my extension.
Usually I have something like:
exten => 55,1,Goto(,yyy,1)
but if 55 does not come across and it appears to be no digits
coming across how do I match that that and jus
On 03/29/16 17:03, Vitor Mazuco wrote:
> Is possible with Telegram?
Telegram does not support voice calls for humans either. It's strictly
an IM system.
They do have a bot API if you want to interface some system with their
messaging system. With it you can send text and also pictures,
recordings
Is possible with Telegram?
2016-03-29 9:39 GMT-03:00, Emiliano Vazquez :
> El 29/03/16 a las 08:29, Steve Howes escribió:
>> I don't think you can. Whatsapp is a closed system.
>>
>> Steve
> And they change your code every day and make it always obfuscated.
>
> https://github.com/tgalal/yowsup/iss
El 29/03/16 a las 08:29, Steve Howes escribió:
I don't think you can. Whatsapp is a closed system.
Steve
And they change your code every day and make it always obfuscated.
https://github.com/tgalal/yowsup/issues/887
Best regards.
Emiliano.
--
___
On 28/03/16 12:46, bilal ghayyad wrote:
Does anyone has information if possible to setup SIP trunk with whatsapp?
How can we let asterisk send and receive calls from whatsapp?
I don't think you can. Whatsapp is a closed system.
Steve
--
Hello;
Does anyone has information if possible to setup SIP trunk with whatsapp? How
can we let asterisk send and receive calls from whatsapp?
RegardsBilal--
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Hello! Thnxs for reading!
I've an IPLAN virtual PBX, that allows me to connect via zoiper or gigaset,
for instance (and it works!)
Connection parameters are:
Authentication Name: Número 11
Authentication password: 12345678
Username: 11
Display name: 11
Domain: hpbx.iplanne
hello everybody,
i want to configure a sip trunk between my system which has asterisk 11.5.1
and a cisco router. this is my scenario:
Freepbx-my system-cisco-routerFreepbx
my system acts like a router. if i set just one codec in dial-peers on
cisco router, every thing is ok and i can
But the phone rings - so its routed - just no audio.
The ringing is SIP signaling. The audio is RTP data. See if the audio
is getting routed with a sniffer. Maybe use one codec that both clients
support.
Adrian Serafini
--
__
I have two machines on the internet. Box A and Box B.
Box A has a SIP trunk to the world, Making calls box A works fine
I have audio to my cell and all works.
I defined a SIP trunk between box B and A. tried to make a call originating
from box B - to box A and then over the SIP trunk to my cell.
On Thursday 31 Jul 2014, James Thomas wrote:
> Is the quality the same incoming from mobile as outgoing to mobile?
It's a one-way trunk (outgoing only).
Anyway, I've now fixed it, with help from the trunk provider. Details to
follow in a separate message.
--
AJS
Note: Originating address o
Is the quality the same incoming from mobile as outgoing to mobile?
On Wed, Jul 30, 2014 at 4:51 AM, A J Stiles
wrote:
> I'm having a problem with a new SIP trunk.
>
> Calls within the UK to fixed lines are fine, but calls to mobiles have
> noticeably poorer audio quality.
>
> I thought it migh
I'm having a problem with a new SIP trunk.
Calls within the UK to fixed lines are fine, but calls to mobiles have
noticeably poorer audio quality.
I thought it might have been a codec issue; we have used G.726 for internal
and external calls (over primary ISDN and GSM). So I tried allowing "a
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mordechay Kaganer
Sent: Thursday, August 15, 2013 5:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP trunk and congestion handling
B.H.
While dialing out i get a lot of AMI responses like this
gt; *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mordechay Kaganer
>> *Sent:* Tuesday, August 13, 2013 10:55 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-us
>
> * *
>
> http://tools.ietf.org/html/rfc3261#section-21.1.2**
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mordechay Kaganer
> *Sent:* Tuesday, August 13, 2013 10:55
om: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mordechay Kaganer
Sent: Tuesday, August 13, 2013 10:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP trunk and congestion handling
B.H.
Ast
half Of *Mordechay Kaganer
> *Sent:* Sunday, August 11, 2013 8:59 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] SIP trunk and congestion handling
>
> ** **
>
> B.H.
>
> ** **
>
> Hello, all. We have a dia
Which version of asterisk are you using ?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mordechay Kaganer
Sent: Sunday, August 11, 2013 8:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP
B.H.
Hello, all. We have a dialer software that runs outgoing telephony
campaigns. We have been using it successfully with PRI cards, now we're
evaluating it's use also with a SIP trunk. Most of the things run perfectly
good without a need to change anything except for dial string, but there's
som
I will test this allso. Thanks.
- Original Message -
From: "Eric Wieling"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, 23 August, 2012 9:32:26 PM
Subject: Re: [asterisk-users] sip trunk failing to register causes sip
Hi,
Thanks. I will try this.
Regards,
John
- Original Message -
From: "Warren Selby"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Thursday, 23 August, 2012 9:24:48 PM
Subject: Re: [asterisk-users] sip trunk failing to register causes si
: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip trunk failing to register causes sip phones
to become unreachable
On Aug 23, 2012, at 10:30 AM, John Cahill wrote:
> I have only seen this problem when using sipgate SIP trunks which actually
> "re
On Aug 23, 2012, at 10:30 AM, John Cahill wrote:
> I have only seen this problem when using sipgate SIP trunks which actually
> "register". If the ADSL connection goes down that the sip trunk uses, the sip
> phones registered locally become unreachable. This happens on any 1.6.x or
> 1.8 versio
Hi,
I have only seen this problem when using sipgate SIP trunks which actually
"register". If the ADSL connection goes down that the sip trunk uses, the sip
phones registered locally become unreachable. This happens on any 1.6.x or 1.8
version of asterisk I've tried. Is there a work around that
Hi,
When *ANY* SIP client (softphone, hardphone, ATA) registers to an Asterisk
server on my LAN and the extension dials out through a remote SIP
provider, the audio is fine for "a while". It then degrades and starts to be
"cracky"/jittery. The extension can call once and again and it will
alway
I have a Grandstream HT-502 device connected to my Asterisk PBX. It is
configured not to place anonymous calls, and from my mostly layman
reading of the invitation that the device sends, it should not be
anonymous. However, the Asterisk PBX sends an anonymous invitation to
our SIP trunk provi
16 dec 2011 kl. 11:29 skrev James Courtier-Dutton:
> Hi,
>
> I have a situation where unfortunately, I cannot use IAX for trunks,
> and need to instead use SIP trunks.
> Is there any way to fit the voice data from more than one simultaneous
> phone call into a single IP packet over the SIP trunk
No.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James
Courtier-Dutton
Sent: Friday, December 16, 2011 5:30 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Trunk
Hi,
I have a
Hi,
I have a situation where unfortunately, I cannot use IAX for trunks,
and need to instead use SIP trunks.
Is there any way to fit the voice data from more than one simultaneous
phone call into a single IP packet over the SIP trunk.
I believe this is possible with IAX trunks, but I don't know ho
Hello,
I am troubleshooting a SIP trunk problem. The system is Asterisk 1.8.5.
The problem is can't make any outbound/inbound. It always get "Number is
not valid 701".
I tried to figure out the reason the call got dropped and couldn't find
out the solution. I noticed that in the SIP debug th
On 02/03/2011 11:41 AM, marek cervenka wrote:
hi,
is there some way to balance accross sip trunks by the number of calls?
example
3 trunks
alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority 3)
alfa have 25 calls now
i want next call terminate to delta. how to find in aste
hi,
is there some way to balance accross sip trunks by the number of calls?
example
3 trunks
alfa(max 30 channels, priority 1),delta(60,priority 2),omega(90,priority
3)
alfa have 25 calls now
i want next call terminate to delta. how to find in asterisk the current
calls number on sip trunk a
you need to set your external IP in the sip.conf to be your public IP after NAT
(assuming your talking over a public network). That way when the sip request
goes out and it sees the IP address your are sending to is outside your
"localnets" it changes the SIP header to use the x.y.z.w IP you se
Hello,
I'm trying to register to my provider sip trunk, I got from him an host IP
(a.b.c.d) to connect to and my provider recognize me based on the fixed IP
(x.y.z.w) he gave me (no need for username and password)
In the sip.conf I add:
[mytrunk]
type=friend
insecure=no
host=a.b.c.d
fromdomain=x
--- On Fri, 5/14/10, Philipp von Klitzing
wrote:
> You were probably caught be the fact that you are using
> extension numbers
> also as SIP user names for your phones (here: 3666). This
> is not a good
> thing to do, better use an alphanumeric username or the
> phone's MAC
> address etc.
Hi!
> Issue solved.
> Looks like all I was missing was one parameter:
> "fromuser="
That's interesting - could be related to this:
http://lists.digium.com/pipermail/asterisk-dev/2006-November/024842.html
You were probably caught be the fact that you are using extension numbers
also as SIP user
Issue solved.
Looks like all I was missing was one parameter:
"fromuser="
Thanks for your time!
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--- On Wed, 5/12/10, Vardan wrote:
> Please change the peers name in any
> server.
> for example:
> server1:
> interboxsip1
>
> server2:
> interboxsip2
If I understand correctly, the peer names can be identical on both servers.
What counts is the "host" entry, I guess. But then again, my SIP
--- On Wed, 5/12/10, Philipp von Klitzing
wrote:
> What are your allowguest= and domain=
> settings in the global section of
> sip.conf?
>
> And which version of Asterisk exactly are you using?
I have no such settings defined yet. Still haven't tried to set them...
Not sure what to put in d
What are your allowguest= and domain= settings in the global section of
sip.conf?
And which version of Asterisk exactly are you using?
Philipp
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New to
Please change the peers name in any server.
for example:
server1:
interboxsip1
server2:
interboxsip2
Vardan
Vieri wrote:
>
>
> --- On Wed, 5/12/10, Vardan wrote:
>
>> please show "sip show users" and sip
>> show peers"
>
> SERVER 2:
>
> sip show users (trimmed to just my sip test trunk):
>
> Us
--- On Wed, 5/12/10, Vardan wrote:
> And "sip show registry"
sip show registry doesn't list anything regarding my "interboxsip" test trunk
because I'm trying to setup a straightforward link such as this one described
here (without user/password):
http://www.panoramisk.com/90/sip-trunk-with-a
--- On Wed, 5/12/10, Vardan wrote:
> please show "sip show users" and sip
> show peers"
SERVER 2:
sip show users (trimmed to just my sip test trunk):
Username Secret Accountcode Def.Context
ACL NAT
interboxsip
And "sip show registry"
Vardan
Vieri wrote:
>
>
> --- On Wed, 5/12/10, Philipp von
> Klitzing wrote:
>
>>> <--- SIP read from 192.168.250.111:5060 --->
>>> SIP/2.0 407 Proxy Authentication Required
>>
>> You need to run the SIP debug on 192.168.250.111 to learn
>> more about WHY
>> the 407 is i
please show "sip show users" and sip show peers"
vardan
Vieri wrote:
>
>
> --- On Wed, 5/12/10, Philipp von
> Klitzing wrote:
>
>>> <--- SIP read from 192.168.250.111:5060 --->
>>> SIP/2.0 407 Proxy Authentication Required
>>
>> You need to run the SIP debug on 192.168.250.111 to learn
>> more
Please look in any conf file that have any relations with sip.conf.
I think you have some records.
And one also, you take this message when calling in both direction?
(server1 call server2 and server2 call server1)
Vardan
Vieri wrote:
>
>
> --- On Wed, 5/12/10, Vardan wrote:
>
>> I have forget
--- On Wed, 5/12/10, Philipp von Klitzing
wrote:
> > SIP/2.0 407 Proxy Authentication Required
>
> Then you have another entry in sip.conf that uses the same
> IP address.
> Delete that, or change the port on one of them, and adjust
> insecure=
> accordingly.
asterisk1 # grep 192.168.250 s
--- On Wed, 5/12/10, Philipp von Klitzing
wrote:
> > <--- SIP read from 192.168.250.111:5060 --->
> > SIP/2.0 407 Proxy Authentication Required
>
> You need to run the SIP debug on 192.168.250.111 to learn
> more about WHY
> the 407 is issued. Have a close look and you are likely to
> unders
--- On Wed, 5/12/10, Vardan wrote:
> I have forget to write for outcall in
> extension
>
> server1:
> [calltoserver2]
> exten => _X.,1,Noop(Call to server2)
> exten =>
> _X.,2,Dial(SIP/interboxserver2/${EXTEN})
> exten => _X.,3,Hangup
>
> server2:
>
> [calltoserver1]
> exten => _
Hi again!
> <--- SIP read from 192.168.250.111:5060 --->
> SIP/2.0 407 Proxy Authentication Required
You need to run the SIP debug on 192.168.250.111 to learn more about WHY
the 407 is issued. Have a close look and you are likely to understand it
right away.
Also: Do not forget the "reload" af
Hi!
> I'm trying option c) which is the simplest.
> used insecure=invite but failed with the same SIP messages.
> Tried also insecure=yes but the same messages show up:
>
> SIP/2.0 407 Proxy Authentication Required
Then you have another entry in sip.conf that uses the same IP address.
Delete th
I have forget to write for outcall in extension
server1:
[calltoserver2]
exten => _X.,1,Noop(Call to server2)
exten => _X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten => _X.,3,Hangup
server2:
[calltoserver1]
exten => _X.,1,Noop(Call to server1)
exten => _X.,2,Dial(SIP/interboxserve
And also please show your settings and logs (without debug)
Vardan
Vieri wrote:
>
>
> --- On Wed, 5/12/10, Philipp von
> Klitzing wrote:
>
>> Either
>>
>> a) set a secret and use that on both sides, or
>> b) look at allowguest= and the default context and maybe
>> the domain=
>> settings, or
>>
Vardan wrote:
> Hello
>
> Server1:
>
> sip.conf
>
> [interboxserver2]
> type=friend
> host=192.168.250.112
> context=callfromserver2
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
>
> extensions.conf
>
> [callfromserver2]
>
> exten => _X.,1,Noop(Call from server2)
> exten => _X.,2,Dial(S
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten => _X.,1,Noop(Call from server2)
exten => _X.,2,Dial(SIP/${EXTEN})
exten => _X.,3,Hangup
Server2:
sip.con
--- On Wed, 5/12/10, Philipp von Klitzing
wrote:
> Either
>
> a) set a secret and use that on both sides, or
> b) look at allowguest= and the default context and maybe
> the domain=
> settings, or
> c) use insecure=invite
Thanks Philipp.
I'm trying option c) which is the simplest.
used i
Hi!
> I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN
> (no NAT, no firewalls).
>
> With IAX2 all's fine but I'm unable to setup SIP. I must be missing
> something obvious.
Either
a) set a secret and use that on both sides, or
b) look at allowguest= and the default
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no
NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing something
obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk
12 mar 2010 kl. 12.01 skrev Klaus Darilion:
>
>
> Am 02.03.2010 13:29, schrieb Magnus Benngård:
>> Hi!
>>
>> Did a setup of 2 peers as Klaus suggested, it worked thx!
>>
>> Has anyone thought about the possibility to add multiple ip/hosts to
>> "host="?
>>
>> I my case: "host=130.244.190.42,
Am 02.03.2010 13:29, schrieb Magnus Benngård:
> Hi!
>
> Did a setup of 2 peers as Klaus suggested, it worked thx!
>
> Has anyone thought about the possibility to add multiple ip/hosts to
> "host="?
>
> I my case: "host=130.244.190.42,130.244.190.46" or
> "host=sip-corporate1.tele2.se,sip-corporat
Hi!
Did a setup of 2 peers as Klaus suggested, it worked thx!
Has anyone thought about the possibility to add multiple ip/hosts to
"host="?
I my case: "host=130.244.190.42,130.244.190.46" or
"host=sip-corporate1.tele2.se,sip-corporate2.tele2.se"
Step 1 could be to send to the first ip/hos
Am 02.03.2010 08:50, schrieb Magnus Benngård:
> Hi,
>
> Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No
> problem to get outgoing calls to work but i have some problems with
> incoming.
>
> Did set "srvlookup=yes" in sip.conf. "Sending" all outgoing calls to
> "sip-corporate.tel
Hi,
Did order and setup a SIP trunk to a Swedish ITSP named Tele2. No problem
to get outgoing calls to work but i have some problems with incoming.
Did set "srvlookup=yes" in sip.conf. "Sending" all outgoing calls to
"sip-corporate.tele2.se" which is either sip-corporate1.tele2.se
(130.244.19
2010/1/26 Peter Childs :
> 2010/1/26 Yves Arikoglu :
>> do you use the
>>
>> qualify=yes
>>
>
> No, If I do it does not work at all.
>
> I've found if I set defaultexpiry to 30 it works fine. and was infact
> working for 30 seconds every two minutes before, It looks like
> sipgate.co.uk are expirin
2010/1/26 Yves Arikoglu :
> do you use the
>
> qualify=yes
>
No, If I do it does not work at all.
I've found if I set defaultexpiry to 30 it works fine. and was infact
working for 30 seconds every two minutes before, It looks like
sipgate.co.uk are expiring there registry attempts very quickly.
do you use the
qualify=yes
option for your endpoints?
y.
Peter Childs schrieb:
> Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash
>
> I've managed to get a basic system set up. and can now take and make
> sip calls over the sip trunk I've got from sipgate.co.uk for testing
> purp
Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash
I've managed to get a basic system set up. and can now take and make
sip calls over the sip trunk I've got from sipgate.co.uk for testing
purposes
Anyway I can make calls fine (if only to the testing line and other
sipgate lines a
Yes, I have qualify=yes
Could this be related to various posts regarding DNS issues?
I doubt I have dns issues because the hostname and IP of the other server is
hard-coded in /etc/hosts
Thanks,
Dan
On Tue, Jul 28, 2009 at 3:38 PM, Ishfaq Malik wrote:
> Hi
>
> Have you tried setting qualif
Hi
Have you tried setting qualify in the sip.conf?
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
Ish
dan julius wrote:
> Hi,
>
> I have configure a SIP trunk between two asterisk 1.4.24.1
> After a while, sometimes a day or two, sometimes only a few hours, the
> SIP connection betwee
Hi,
I have configure a SIP trunk between two asterisk 1.4.24.1
After a while, sometimes a day or two, sometimes only a few hours, the SIP
connection between the two servers is lost.
'sip show peer status' shows the peer is unreachable.
'sip reload' resolves the problem, but I'm wondering if there
CNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
From: mlecu...@gmail.com
Date: Wed, 27 May 2009 14:17:23 -0300
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP Trunk groups
Hey all,
I have 2 GSM to Voip gateways and probably we will grow
up to 4 m
I've improved this since this revision, but now a days I don't use limited
systems. But my code has been used in places that need 100 concurrent
outgoing lines.
[macro-which-line]
exten => s,1,set(TRIES=0)
exten => s,n(nextone),set(TRIES=$[${TRIES} + 1]) ; increment TRIES by 1
exten => s,n,set(DI
AFAIK, unfortunatelly it's not the same as with ZAP channels where you can
group multiple lines together.
I ended up using slightly modified superdial macro:
http://www.voip-info.org/wiki/view/Superdial+macro.
if you add new gateway it's not necesarry to edit the macro, just add new
line in dialing
Hey all,
I have 2 GSM to Voip gateways and probably we will grow up to 4 more
gateways. I already created a macro to make failover happen between
gateways, but can imagine that everytime I add a new gateway I will need to
modify the macro. The initial intention of this macro was to failover
betwe
He all,
I have 2 GSM to Voip gateways and probably we will grow up to 4 more
gateways. I already created a macro to make failover happen between
gateways, but can imagine that everytime I add a new gateway I will need to
modify the macro. The initial intention of this macro was to failover
betwee
users@lists.digium.com
Subject: Re: [asterisk-users] SIP trunk with > 250 lines
"Danny Nicholas" writes:
> Okay - I'm not shooting from the hip here. The driver in question is a
> Intel E1000 on a Poweredge 1650. If you visit the Digium site and do
other
> googling, you w
"Danny Nicholas" writes:
> Okay - I'm not shooting from the hip here. The driver in question is a
> Intel E1000 on a Poweredge 1650. If you visit the Digium site and do other
> googling, you will see that there is a specific issue with asterisk and this
> hardware/driver combination. I'm not r
Danny Nicholas wrote:
> Okay - I'm not shooting from the hip here. The driver in question is a
> Intel E1000 on a Poweredge 1650. If you visit the Digium site and do other
> googling, you will see that there is a specific issue with asterisk and this
> hardware/driver combination. I'm not really
On 25/03/2009 10:54 a.m., Danny Nicholas wrote:
> I downloaded the newest E1000 driver from the Intel site and tried it on a
> 1550 and 1650 with no joy. So this isn't an attack on Dell, just a
> verification of information I found and was trying to pass on to the
> questioner. It could just as e
rcial Discussion
Subject: Re: [asterisk-users] SIP trunk with > 250 lines
On 25/03/2009 10:05 a.m., Danny Nicholas wrote:
> It's actually a E1000 on Any POWEREDGE. If yall want a rukus, I can trash
> Dell all day. That's not really what I had in mind though.
Hmmm, I've als
On 25/03/2009 10:05 a.m., Danny Nicholas wrote:
> It's actually a E1000 on Any POWEREDGE. If yall want a rukus, I can trash
> Dell all day. That's not really what I had in mind though.
Hmmm, I've also had problems with the e1000 driver in the past but not
on Dell - I seem to remember reading so
wards
Sent: Tuesday, March 24, 2009 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP trunk with > 250 lines
>>> On Tue, 24 Mar 2009, Danny Nicholas wrote:
>>>
>>> Using conference rooms will increase your bandwid
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