similar to: Maximum retries/no reply to our critical packet

Hello All, I am trying to setup a small system where Nextone Softswitch will send traffic to Asterisk and then terminate on Grandstream GXW410x IP Analog Gateway but for some odd reasons the call are flashed back from Grandstream to Asterisk and creating a Black loop... I did follow the instructions provided by Grandstream support but it doesn't seems to be working...

Hello All, "safe_asterisk" just sent me an email saying "Asterisk on bill exited on signal 11. Might want to take a peek.". Looking at the /var/log/asterisk/message doesn't show me anything... This is a fresh installed Asterisk 1.6.2.6 on Ubuntu 9.10 (64-bit) and it is routing calls from Nextone MSW Softswitch to VPS Softswitch... Any reason why Asterisk died?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U)

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

I have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c for seqno 1620 (Critical Response) Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up call

Hello My asterisk is receiving calls from OpenSER but all calls hangup in 20 seconds. This only happens because Im using Asterisk2Billing's AGI (without A2Billing it doesnt hang up). does someone knows whats the problem?? Here is my Asterisk debug: (xxx.xxx.xxx.xxx -> the phone's IP) Apr 10 02:03:02 WARNING[6996]: res_musiconhold.c:508 monmp3thread: Unable to spawn mp3player Apr

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

I have a situation here where a user has an AAstra 480i phone, which function corectly. The phone is behing a nat-router (a linksys wrv200 for it's VPN point to point facility). The phone is plugued in a port wich has qos enabled. And when the user places a call, sometimes (not always), we get this in the console : [May 8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum

have searched this list and others, and see other pepole having this issue. However, I have not seen how to fix it. Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Maximum retries exceeded on transmission 778f89593967725f0abe40eb1752504c (at) 10.10.206.53 for seqno 1620 (Critical Response) Sep 12 18:52:36 WARNING [4620]: chan_sip.c:1835 retrans_pkt: Hanging up 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!

I often get a Maximum retries error while making outgoing calls. Why does this happend? Most of the time a reload solves the problem, but not all the time? What to do? Aug 26 09:52:46 WARNING[6613]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 32166-1B75-151E-E6DA-E37AEEAA2882@10.100.4.252 for seqno 1 (Non-critical Response) Regards, Arne Morten.

Hi, i've just got 2 grandstream phones and when I try to connect them with * I get the following: -- Playing 'demo-abouttotry' (language 'en') WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 02da8d3d-a8de-c7e1-5f07-c032ab23d909@10.1.1.179 for seqno 59134 (Response) I've seen there was some discussion on this already but i

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",

Hi all, this message occurs if i was connected or not: WARNING[213006]: File chan_sip.c, Line 432 (retrans_pkt): Maximum retries exceeded on call 0b03e0c6189a769b54e49eb471f32454@172.20.23.150 for seqno 102 (Response) If i was connected, the call will be disconnected after a few seconds. What does it means ? I don't see anything to configure like Max retries.... Thanks for help, Thomas.

Hello I'm tryng to install Asterisk and by now I got a first congfiguration working (0ne PBX box and 2 X-lite phone communicating with each other) The problem now is that I keep this annoying message every time: WARNING[5126]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call 43c827415f83584c0c9dc15b03ed6924@10.1.1.1 for seqno 102 (Request) Do you have a

There is no static interval. But i found out that it was my IP-Phone Service Provider that was having serviceproblems today. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Giorgio Incantalupo Sendt: 26. august 2005 11:33 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users]

Thank`s all, Chris Ziomkowski wrote: > If you only want to use H.323 with Asterisk, you should configure it as an H.323 gateway. > Why are you trying to set "softswitch"? I was asked by a costumer, because he could not use a asterisk as a softswitch in the Nextone configuration, so I`m looking for the difference in asterisk configs files. Thank`s a lot for you help...

First of all, I'd like to send a big "thank you" to all the folks who have helped me get this far. Now on to the next problem. Here's my current network setup: The Big I ---+--- FreeBSD FW --- * (10.0.0.253) ---- PC (10.0.0.1) | +--- Laptop (public IP) natd is set up with the following rules: redirect_port udp 10.0.0.253:10000-20000 10000-20000

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As an example....if we have a call that: 1. originates via PSTN line to one of our local DID's in Seattle 2. comes into our Asterisk server in Los Angeles or Denver 3. is routed by Asterisk for termination back to a different Seattle PSTN ....and if our VOIP call termination provider requires (in order to get their best rate) all calls to go through their Nextone