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Specs: Asterisk 1.4.22 running behind a SonicWall (transparent mode) with a public IP address. We have our phone system setup as 172.16.2.x that connect through the SonicWall to Asterisk. Incoming calls work flawlessly and we no longer get one-way audio. We are only using SIP (3 trunks now, instead of 2) and having all 3 in use is not an issue. Problem: Make a call on a Polycom 320 IP phone to

I'm not getting inbound audio from bandwidth.com. Their engineer said the invite that they're sending me looks like this: INVITE sip:+15129616808 at 67.198.16.18:5060;transport=udp SIP/2.0. Record-Route: <sip:216.82.224.202;lr;ftag=VPSF506071629460>. Record-Route: <sip:4.79.212.229;lr;ftag=VPSF506071629460>. Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bK6314.4f7b5d05.0. Via:

Hello All, I have Cisco 7960's, Cisco 2950 Switch. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone is disconnects at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone seen this? Thanks for the help,

Quick update on my issues, Voicemail doesn't pickup also. It just drops the line.. Thank you Chris Tuska ------------------------------ Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the

hi, let me explain in detail, what i have configured and what is happening now: 1st router w724v (Deutsche Telekom AG): - port forwarding, everything to destination port 51000-55999 to device with ip 192.168.2.50 (interface of 2nd router) 2nd router Bintec RS353j): - configured NAT, everything to port 51000-55999 to device 192.168.3.99 (same ports) other direction is totally open. I

Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone it then disconnects the call at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone

For reasons unknown to me, SER and subsequently a Microsoft Live Communcations Server 2005 seems to have problems, matching a SIP ACK request from asterisk to the ongoing SIP transaction, I have attached the complete log, but the essential lines are: 13(2894) DEBUG: RFC3261 transaction matching failed 13(2894) DEBUG: t_lookup_request: no transaction found 13(2894) SER: forwarding ACK

I have the following scenario: PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) -> Asterisk 1.2.16 (203.89.nnn.nnn) When an incoming call is received, often (but not always) Asterisk repeatedly sends a SIP 200 OK message and eventually hangs up the call. sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

On about 25% of inbound calls to a ring group, picking up any one extension as it rings results in dead air. Some details regarding my VoIP network to make the following logs more readable: 192.168.7.130 resolves to the trixbox host. 192.168.7.135 resolves to endpoint 812. 192.168.7.137 resolves to endpoint 811. 192.168.7.138 resolves to endpoint 810. 192.168.7.139 resolves to endpoint 813.

Hello all: For fun, I am learning about Asterisk, and trying to get Asterisk working at my house. I installed Asterisk@Home. It seems to be functioning fine. I installed a couple of softphones, and have them registered with Asterisk. I actually work for a CLEC, and I have registered my Asterisk box with SER (which I don't begin to understand yet) at the office. In order to try to

Hi all. Just as a quote note, can I thank everyone on this list. I find my self finding pretty much every answer I am looking for on here. And a big thanks to all thoughs helping us out. Mass Respect :) Ok, I'm using a SIP provider (SipGate UK) to do my international dialing etc, working great from extension 8 on phones. However some more friends/contacts have started using SipGate also, and

Hello, I was interconnecting Asterisk (v1.0) with a strict router (ie, no ;lr in routes) and I think I found a bug in the way Asterisk prepare new requests inside a dialog. I'm sending some captures (ngrep) along with my comments. This is a 200 OK (INVITE) received by Asterisk ========================= U 2005/02/10 16:41:55.065538 143.173.202.82:5060 -> 143.173.202.83:5070 SIP/2.0 200

I have setup my asterisk box using freepbx. I can call extension and make outbound calls. the outbound calls drop between 10-30sec. we are using bandwidth.com and they have logged our call. below is your bad followed by what they say is a good call. I can't figure out where the problem is on your end. I know we are missing some stuff at the bottom but I don't know where to start.

OK, here goes my next problem. I have puchased a DID which I can connect to via SIP I have been given the following details: Username: uka1xxxxxx Password: 1000xxxxxx Server: brxxxx.net:5160 My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT) The other end is a Cisco AS5300 (NO NAT) I can register with the Cisco with no problem. When I dial the DID it sends the call to my asterisk

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

I have a DID but for some reason is not working in asterisk-1.6 The same sip connection in asterisk-1.4 is working OK, but it doesn't work with asterisk-1.6 Here is my sip.conf section: ... [actio-out] type=friend secret=password user=48746612254 username=48746612254 fromuser=48746612254 authname=48746612254 callerpage=48746612254 fromdomain=sip.actio.pl host=sip.actio.pl insecure=very

Hi, I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing

Hello. I'm trying to use Asterisk in combination with SER, to make the routing proccess to my PSTN-Gateways. I made a simple test defining some extension in my extension.conf, when i made a call my SER (SIP) Server forward the call to Asterisk, this proccess is ok, but when the call is answered i see an INVITE going out from Asterisk to my SER Server, this invite is then passed to my

Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? Below you can see the output of the asterisk monitor. <--- Received SIP request

Hi all, I try to make a call from my Openser(SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip