Displaying 20 results from an estimated 1000 matches similar to: "Calls being dropped"
2009 Dec 24
2
1.6 Troubleshooting help
Hi,
How would I go about troubleshooting this:
[Dec 24 07:15:11] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum
retries exceeded on transmission a50346a4-bfdc32ed at 192.168.1.95 for
seqno 101 (Critical Response) -- See doc/sip-retransmit.txt.
[Dec 24 07:15:12] WARNING[5228]: chan_sip.c:3397 retrans_pkt: Maximum
retries exceeded on transmission 90bd2c4d-aaaec88 at 192.168.1.95 for seqno
101
2004 Jan 24
4
retrans_pkt: Maximum retries exceeded on call
Hey,
I'm getting an odd message in my logs, and have'nt been able to find much information on it:
Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call 6010532c6fedf9be383872e07e4be70c@192.168.1.2 for seqno 102 (Request)
I'm running asterisk with a Cisco 7960G
If anyone know's why i'd get this.....Any help would be appreciated!
2004 Mar 16
6
Maximum retries exceeded on call
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-
000A95DA04DA@192.168.1.152 for seqno 48221 (Response)
== Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7'
Mar
2017 Jan 28
4
Asterisk 13.13.1
On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4h07 at gmail.com> wrote:
> What did you exactly upgade ? Asterisk only ? Asterisk and OS ?
> How did you installed Asterisk 1.8 and 13 ? From source or from package ?
>
> I would be curious to see what would happen after downgrading back to 1.8.
>
> 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.cruz at gmail.com>:
>
>
2008 Oct 28
2
Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
Hi All,
I've looked through the archives and tried several variations in Google, and I haven't found anything on-point... So I'm hoping someone here may be able to help this relative Asterisk neophyte shed some light on an issue:
I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G phones and an AEX804E card (4 FXO, hardware echo cancellation).
The server and all
2004 Mar 31
2
SER Asterisk problem
Hi All.
I'am using Asterisk with SER. I can make call between two internal VoIP
gateways or from na internal to external VoIP gateway. But when I get a
external call, this call hang ups 5 seconds after and I reveive the
following messages
*CLI> -- Executing Dial("SIP/16008-3d17",
"SIP/16007&SIP/16006|20|tr") in new stack
-- Called 16007
-- Called 16006
2003 Sep 24
10
SIP / GrandStream Configuration
Hi there!
I installed the BudgetTone (GrandStream) on my LAN without any problems.
Then, I moved it to another location using a D-Link NAT.
I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address
of the BudgetTone.
When I receive a call on my Asterisk, it would ring my FXS as before.
However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in
the log).
The
2004 Nov 30
2
Dual NAT for SIP
Hi,
My installation at home use two NAT translations before it reaches the linux box where Asterisk is running on.
I use DSL with a Wireless router which fwd all packets to an Windows 2003 box an this windows box it NATing the UDP and RTC packets to my linux box.
If I try to connect to it from outside I get this error :
Nov 30 22:19:02 WARNING[1106250672]: chan_sip.c:673 retrans_pkt: Maximum
2007 Aug 06
1
sip issue with one way audio
I am getting this error
[Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum
retries exceeded on transmission 8f68421-22821e1e at localhost for seqno
102 (Critical Response)
[Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging
up call 8f68421-22821e1e at localhost - no reply to our critical packet.
any Ideas?
Jason
2003 Dec 11
2
SIP retries
Is there a way to increase the number of retries or the time to help
with this?
WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries
exceeded on call 0ea2761d6a82fa49221f547c739bde18@192.168.0.200 for
seqno 103 (Request)
WARNING[40966]: File chan_sip.c, Line 462 (retrans_pkt): Maximum retries
exceeded on call 0ea2761d6a82fa49221f547c739bde18@192.168.0.200 for
seqno 103
2009 Apr 13
2
retransmision error con asterisk 1.4.24.1
se?ores alguien le ha presentado este problema al acceder al voicemail
o al hacer una llamada a la pstn
1940> Playing 'vm-received' (language 'es')
-- <SIP/111-08d91940> Playing 'digits/yesterday' (language 'es')
-- <SIP/111-08d91940> Playing 'digits/at' (language 'es')
-- <SIP/111-08d91940> Playing
2004 Jan 27
1
Cisco 7960 Problems
Has anyone ever seen these errors generated by a cisco 7960? none of our other brand phones seem to generate these erros:
Jan 27 21:54:07 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie
s exceeded on call 000ded24-d7000024-5d2ca17a-29c81cf4@65.204.176.54 for seqno 1
01 (Response)
Jan 27 21:54:08 WARNING[-1147556944]: chan_sip.c:486 retrans_pkt: Maximum retrie
s exceeded on call
2013 May 15
3
Cut offs on outgoing SIP calls
Hello everyone,
I've suffering cut offs after 6 or 7 seconds a call is answered, incoming
calls are working fine, but outgoing ones show the gollowing messages when
are being dropped:
[2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt:
Retransmission timeout reached on transmission
ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical
Response) -- See
2007 Dec 11
1
Asterisk not sending 200 OK
We're trying to get a SIP peer going between our asterisk box and our
provider. It should then ring our phone.
The call does come in and it does execute the extension in the dial
plan. But the provider says they never get a 200 OK back and therefore
they send another INVITE and then after a few seconds drop the call.
Here's our setup:
sip.conf
[ngt-trunk]
type=peer
qualify=yes
port=5060
2004 Jul 31
3
MGCP & Cisco ATA 186 Help
Does anybody has the expirience configuring Asterisk with Cisco ATA 186
MGCP firmware ?
I have Cisco software v3.1.1 atamgcp (Build 040629A)
Asterisk 1.0-RC1
On ATA i only put domain test.
mgcp.conf looks like this
[test]
host = 192.168.195.55
context = default
line => aaln/2
line => aaln/1
Asterisk CLI shows this:
Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485
2004 Apr 29
3
Dropped calls -> reproducing scenario
So I think I am able to reproduce the dropped call scenario.
Here is what I do to get a dropped call:
1. Call 1-800-tmobile
2. Go true their IVR and get connected to the customer service IVR
3. Enter my number and SSN
4. press 0
5. Then the audio please hold starts. After about 2-4 seconds the call
gets dropped. (fast busy tone)
The time on my phone will stop running (call time) and I will get
2004 Jan 08
5
Dialing the Phone from OS X Address Book with AppleScript, XML-RPC, PHP and Asterisk
I run an Apple OS X workstation and I've got a server on the same LAN
that's both a webserver and an Asterisk PBX.
I wanted to be able to originate calls in the OS X Address Book
application, and have Asterisk dial them and connect them to the phone
on my desk.
I've assembled a system that uses AppleScript to connect, via XML-RPC,
to a web application that, in turn, connects to
2009 Oct 15
4
Calls hang up after 20 seconds
Hello.
I have a problem with Asterisk, sometimes it hangs up an external call after 20 seconds, apparently without any reason.
The call comes from a SIP server hosted from EuteliaVoIP, many peers rangs and one of them answer, the call ends itself after 20 seconds from the answer.
I've tried many configuration in sip.conf, but no one solved the problem.
Log from /var/log/asterisk/messages:
2018 Aug 29
3
getting invites to rtp ports ??
I'm getting invites to very high ports every 30 seconds from a
particular ip address:
Retransmitting #10 (NAT) to 5.199.133.128:52734:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734
From: <sip:37120116780191250 at 67.80.191.250>;tag=1872048972
To: <sip:3712011972592181418 at 67.80.191.250>;tag=as3a52e748
2004 May 04
3
Maximum retries exceeded problem...
Searched the archives thoroughly...
Can't find this specific problem...
Simple setup with Asterisk on RedHat. No voice cards in the box, 2 SNOM 200
phones...
Phones seem to work well, can leave VM, Message Waiting Indicator lights up
but when I try to retrieve messages the call terminates and the following
happens:
-- Executing VoiceMailMain("SIP/520-a25e",