similar to: from asterisk 1.6 to 1.8 - sip trunk unreachable

hi, my asterisk is managing around 500 sip peers, and everytime I do a "sip reload" many sip-peers get "LAGGED" and some get even "UNREACHABLE". Any suggestions ? cu, florian -- florian meister EMAIL: florian.meister@medienhaus.at TELEPHONE: +43 5572 501 134 FAX: +43 5572 501 97134 ADDRESS: gutenbergstrasse 1 6858 schwarzach

Hello, I recently upgraded from asterisk 1.4 to 1.6. I am using the same SIP settings in sip.conf in this version also. I am facing a problem when a SIP client makes a call. When a SIP client registers to asterisk its status shows 'OK' and it is able to receive incoming calls. But as soon as this client make a call, its status becomes 'UNREACHABLE' and it cannot receive any

Dears Friends, I currently have 16 Cisco SPA525g phones with a SPA500s (Attendant Console) connected to each phone. All of the 16 phones, have their Attendant Console configured the same way, where they are subscribing to each of the 16 phones. When I power on the switch, where all the phones are connected to, I then get 16 registers, and 256 subscriptions (16 * 16) happening at the same time.

I get these on a consistant basis for most of the providers I have configured. Some less than others. I even get it from my phone at home to my * box at our data center. What I'm confused about is why it always shows the ping times at right around 2000 ms. That just can't be right. It's always right at 2000 ms. Never less or more by more than 100 or so. May 12 17:42:23

Hi All, I have some problem with Asterisk 1.8 and DIal() to SIP unreachable friend. My dialplan: exten => _XXXX,1,Dial(SIP/${EXTEN},60,rt) Now, when I Dial extension 1050, and there is no 1050 peer registered I got: [Dec 18 22:51:04] WARNING[2307] chan_sip.c: sip_xmit of 0xc2e1330 (len 843) to 0.0.4.26:5060 returned -1: Invalid argument In 1.6 there was no problem, I have got Channel is

Just starting to play around with AsteriskNOW, and tried running conary: Error reading config file http://rmirror.digium.com/conaryrc: No route to host Anyone know if this is just a temporary error? Being rather more familiar with RPM-based system than rpath/conary, I am wondering whether I might just as well put together my own systems using CentOS, Asterisk 1.4 and Asterisk-GUI. At least I

Dear Royce; Did ur problem resolved? Because now I am facing same problem. It look like that it happens with IAX trunk only, but does not happen with IAX endpoints that registering (as trunk does not register, it sends the call directly). My initial analysis that one of the following can help to let the trunks talk: if there is an IAX endpoints registering to the machines, then trunk become

Hi, my asterisk server has a strange behavior when all trunks are unreachable (for example due to internet connection), it doesn't accept registrations from any internal extension, and, obviously, I can't do any internal call. On the log I see only lines like: [2013-09-30 01:35:04] NOTICE[1827] chan_sip.c: Peer 'xxx' is now UNREACHABLE! Last qualify: 36 and then [2013-09-30

Hi! from 2 days I'm trying to run hylafax server and iaxmodem with Asterisk 1.8.5. I have 2 computers in the lan, one is the Asterisk PBX and the other is the server with hylafax and iaxmodem installed. In Asterisk I set up an IAX trunk in this way: ___________________________ iax.conf [iaxmodem] type=friend context=outgoing-fax disallow=all allow=ulaw username=iaxmodem secret=password

I'm having some trouble with Polycom Soundpoint phones. I have had good luck deploying them on a local network, but now I've tried putting some in place which access their * server across the network. The * server is on a public IP and the polycoms are behind a NAT on a cable modem broadband connection. Every so often I get: May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now

On Mon, Jun 12, 2017 at 09:51:53AM +0200, Reindl Harald via samba wrote: > Am 12.06.2017 um 09:41 schrieb Mike Brown via samba: >> On Mon, Jun 12, 2017 at 09:28:20AM +0200, Reindl Harald via samba wrote: >>> Am 12.06.2017 um 09:03 schrieb Mike Brown via samba: >>>> On Mon, Jun 12, 2017 at 01:53:10PM +1200, Andrew Bartlett via samba wrote: >>>>> On Sun,

i have this configuration , An Asterisk server connected to my private LAN 192.168.10.0/24 when i do port forwarding for port 5060 so that i make a call from Internet into Asterisk wireshark show the message "destintion port unrechable" i configured sip.conf for "nat=yes" and "qualify=yes" and "externip="my public IP" did i forget some other ports

I have a network of offices using Asterisk that are connected via IAX2 trunks. The trunks work great for a day or two then for no reason at all one end of the trunk will become UNREACHABLE while the other end is still connected. The only way to fix the problem is to shutdown Asterisk completly then start it backup again. The end that dies is not always the same, some times it is server A and some

Am 12.06.2017 um 10:00 schrieb Mike Brown via samba: > On Mon, Jun 12, 2017 at 09:51:53AM +0200, Reindl Harald via samba wrote: >>>>> Damn firewall. By default, Samba isn't allowed to connect. Found it by >>>>> using wireshark to look at the packets and that gave me the clue >>>> >>>> no need for wireshark - normally one does simply

Hi, I'm experiencing a strange behavior on one of my asterisk servers. When I make a sip connection (sip.conf) between the 2 boxes using the primary interface it works fine. When however use an aliased interface (ethX:Y) it fails. from sip show peers: ast-rem-vpn 192.168.3.253 A 5060 UNREACHABLE when I use the ethX interface directly, it works fine. Did anyone

Hi, I'm having a problem trying to register sip trunks. I using asterisk 1.4.39.1, freepbx 2.5.2 in centos 5.5 and I'm trying to configure several sip accounts from my provider. The accounts are individually configured as sip trunks. With only one account everything is ok, it registers and I can make and receive calls. My problem is when I try to put more accounts, it seems to start

I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk the definitive guide", 4th ed. While I don't have the page handy, I was reading the suggestion to try SIP to SIP before proceeding to outside connectivity. I'm aware that SIP trunking is a construct, but am, obviously, learning the system. What I'd like to do is from the CLI "ping"

Name/username Host Dyn Nat ACL Port Status 2011/2011 10.1.1.10 5071 UNREACHABLE 2010/2010 10.1.1.10 5070 UNREACHABLE 2009/2009 10.1.1.10 5069 UNREACHABLE 2008/2008 10.1.1.10 5068 UNREACHABLE 2007/2007

I have logs full with this messages... I must have qualify turned on, because phone is behind firewall, main problem si, that phone is each hour about one hour unavailable! :'( I tried to modify minexpiry/maxexpiry sip.conf timeouts, but nothing help me. I'm using latest firmware 8.4 in phone, will be better to downgrade? to what version? (latest asterisk 1.4branch) [Dec 7 00:36:56]

Hi list! I have a problem where SIP packets sent by Asterisk do not hit the wire, and I don't know what could cause this. I'm running Asterisk 1.8.28_cert5 with full SIP debug. At the same time, I'm doing a tcpdump of the traffic on the network interface. I can see in the SIP debug log that asterisk is sending packets. Most of the time, I can see those packets in the tcpdump,