similar to: SIP Phone Unreachable

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

I am trying to create an IVR system that uses both POTS and IP phones and I have a few problems that I encountered with the IP SIP phones (Grandstream Budge Tone 102). 1. When a user hits the hook fast enough, the user can create multiple IVR connections that gives the appearance of an echo that is phased a few seconds apart. The way to reproduce this is by hitting the hook fast and furious. The

I have a situation where we have an asterisk box that is extending several Mitel PBX extensions to some cordless SIP phones (Cisco WIP310). Everything works great, except when the cordless phone walks out of range of one access point and into range of another (cisco 1100 series APs). I have another post to the list asking about how to speed up the handoff, and keep the call active while

I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used

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

Dear, Kindly guide with the 2 issues mentioned below *#1* - *Host unreachable 0 last qualify 0 (only in 3G**)* I am trying to use SIP client over 3G. It registers and call can be initiated from the client but it can't receive call; cause *asterisk sever *marks it as unreachable immediately after registration. "[2014-10-08 14:32:47] NOTICE[1610]: chan_sip.c:29596 sip_poke_noanswer:

Greetings, I have a home network setup with a broadband router sharing a connection to a MDK Linux machine and an XP Pro machine. The XP box is setup as master. At first I could access the xp machine from the linux box only. I changed the smb.conf file and uncommented the [tmp] section and could access that directory from the xp box. Now I can't mount the shared directories on the xp

Is anybody using Nortel IP Phone? I have (second hand) Nortel i2002 phone and when it boots I get: DHCP server unreachable F/W version: 0604D9C My setting: DHCP? [0-No, 1-Yes]: 1 DHCP: 0-Full, 1-Partial: 0 Can any body suggest how to troubleshoot it? -- #Joseph GPG KeyID: ED0E1FB7

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

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

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 Everyone, I'm new to * and I'm trying to setup a small configuration of SIP clients. Eventually when I get this working I plan on expanding with a Digium developers kit to add analog phones and PSTN access. My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both peers seem to register with * but I cannot call to one another. When I dial the associated extension, the

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

Hey, I just started trying to use the qualify=yes option on my Cisco 7960 SIP phones. Of the 13 I have, 2 of them seem to loose their registration with asterisk on a regular basis. I see lots of these lines: -- Registered SIP '3030' at 62.74.107.1 port 58825 expires 60 in my console. But I only see them for 2 extensions. Never see them for the other 11. All 13 phones have the exact same

I've noticed a little problem with my setup. I've been using a flaky version of X-Lite for testing, and it tends to crash every few phone calls. Since I'm just using it for testing, I don't really care, but it's exposed a problem: when the SIP client goes away, their calls are left in limbo. I just had to soft-hangup a multi-hour outgoing call that had belonged to my

Hi, I am trying to implement call forwarding on the event that my ATA was not reachable to Asterisk, whether due to registration timeout, network interruptions between the ATA and Asterisk, or simply because the network on which the ATA resides in unreachable for any reason. I there a way of implementing such a feature in Asterisk? I have implemented CF unconditional, and CF on busy, CF on

Hi, I have my asterisk running directly plugged on an static public IP accessible by Internet. I try to contact an Cisco VoIP system on another static public IP address but asterisk return me 403 - Forbidden. When I debug sip flow, it tells me : Network Unreachable. I'ts certainly an error in my sip.conf but I can't find where it is... I'm sure about IP addresses. SIP.conf

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

Hi all, After running for months without issue I've got a situation where incoming SIP calls to my asterisk server are failing while outbound calls appear to be working as expected. The server is a gateway between my home LAN and a broadband cable connection with a dynamic IP. The gateway runs FreeBSD 7.1 and Asterisk 1.6.0.15 (built from ports) and registers to my ISTP no problem. Outgoing

Hi! I recently upgraded Asterisk from version 1.6.2 to 1.8.5 Now about every 10 minutes all SIP TRUNKS becomes UNRECHABLE for a few seconds or minutes after become LAGGED and later become OK. I have no idea of the cause of this problem. With the version 1.6.2 all runs perfectly. I can't say more because I have no idea where the problem is. Any suggestions? thanks -------------- next