I am confused due a side effect produced in my * installation.
It consists of 1 Sangoma A101 E1/lSDN PRI card connected to Telmex
service
16 analog phones thru SIP enabled SP5004 Micronet gateways 4 SIP hard
phones.
Everything in a local network/no natting.
We are processing nearly 2000 calls/day
Using SIP? IAX?
One way sound is usually a SIP and nat/firewall problem, make sure ports are
forwarded.
Steve
- Original Message -
From: Peter Ankerstål [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 12, 2005 10:39 PM
Subject: [Asterisk-Users] Maximum
is usually a SIP and nat/firewall problem, make sure ports are
forwarded.
Steve
- Original Message -
From: Peter Ankerstål [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 12, 2005 10:39 PM
Subject: [Asterisk-Users] Maximum retries exceeded on call.
I
I have set up a asterisk-server behind NAT and peers to another asterisk
and uses that one for outgoing calls. I have som clients on my asterisk
and they could register to it well over internet. Not a problem. But when
they try to call me the asterisk-server tells me this:
Oct 12 23:21:38
has somebody an advise.
Do I need to provide more information?
Regards
Michael
Michael Häberle wrote:
Hi,
I phone with phpagi and/or x-pro.
Sometimes I get this warning in the asterisk-console:
maximum retries exceeded on call.
I noticed when this message shows up, asterisk hangs up the call
Hi,
I phone with phpagi and/or x-pro.
Sometimes I get this warning in the asterisk-console:
maximum retries exceeded on call.
I noticed when this message shows up, asterisk hangs up the call (even
when i'am in the middle of a call, according to our employess)
When they restart x-pro it seems
I see this message in my asterisk log sometimes. Can someone explain to me
what this means and how to correct the problem?
May 17 13:38:33 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call 795fcf0c6
[EMAIL PROTECTED] for seqno 18950 (Non-critical Response)
May 17
We see this when person A forwards their phone to person B, who has
forwarded their phone to Person A.
so A-B-A
or A-B-C-A and so on and so forth :)
No matter how many times I tell 'em ...
Julian
Joel Jn-Francois wrote:
I see this message in my asterisk log sometimes. Can someone explain to
me
None of our phones are being forwarded unless the phones are being
forwarded unknowingly.
We see this when person A forwards their phone to person B, who has
forwarded their phone to Person A.
so A-B-A
or A-B-C-A and so on and so forth :)
Joel Jn-Francois wrote:
I see this message in my
I see this message in my asterisk log sometimes. Can someone explain to me
what this means and how to correct the problem?
May 17 13:38:33 WARNING[13503]: chan_sip.c:694 retrans_pkt: Maximum retries
exceeded on call 795fcf0c6
[EMAIL PROTECTED] for seqno 18950 (Non-critical Response)
May
i am server with ser(iptel) and asterisk.
the prosess is:
the xlite the connect to ser, the ser redirect to asterisk and call for x100p
after the two minutes the call, the call is cut.
the error is:
Asterisk Ready.
*CLI -- Executing Dial(SIP/sorcier.com.pe-41000490,
Zap/1/499732) in new
Hi:
I have a asterisk server that shows the following Warning
Jan 24 10:23:37 WARNING[1116941120]: chan_sip.c:673 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 102 (Non-critical Reque
If it show on the midel of the call rhe call will be droped or if it show at
the
Any idea why I keep getting this on my Asterisk console? This is coming from
a Grandstream 101..
No NAT or anything, clean connection between phone and Asterisk on same
subnet via 100mbit.
***
Oct 26 15:19:52 WARNING[180235]: chan_sip.c:681 retrans_pkt: Maximum retries
Eric C. Snowdeal III wrote:
after registering the phones correctly and receiving a 200 o.k.
message i can connect to other registered softphones and pstn
endpoints [ via an voicepulse account ], but after making the initial
connection, i can't hear any sound and i get disconnected after
i'm new to asterisk and am having trouble placing outbound calls. i
Bug Grandstream so that they finally fix their buggy software.
The GS phone sends occassional SIP packets to port 0, not to port 5060, as
tcpdump or (better) ethereal will show you.
There's a page on this at voip-info.org.
Holger Schurig wrote:
i'm new to asterisk and am having trouble placing outbound calls. i
Bug Grandstream so that they finally fix their buggy software.
The GS phone sends occassional SIP packets to port 0, not to port 5060, as
tcpdump or (better) ethereal will show you.
There's a page on
i'm new to asterisk and am having trouble placing outbound calls. i
know this topic has been discussed ad nauseum in the past [1] , but i
can't seem to find a workaround and i'm wondering if my newbie-ness is
getting the best of me.
after registering the phones correctly and receiving a 200
Running * with default config files except for sip.conf. Any call made is
dropped 5 seconds after connection, with the following messages:
Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call 6C94C1B1-77C4-11D8-91FB-
[EMAIL PROTECTED] for seqno 48221
check NAT setting try taking it out of sip.conf, that worked for me
Barry
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On Tue, Mar 16, 2004, Barry Fawthrop thus spake:
check NAT setting try taking it out of sip.conf, that worked for me
Nope. My bad-shoulda said up front that I've tried both with and
without nat=yes in sip.conf, no difference in symptoms.
Regards,
Ed Hintz
[EMAIL PROTECTED]
check context perhaps try include in the extensions.conf
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