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  • SIP and NAT routers?

    - by OverTheRainbow
    Hello SIP was not built with NAT routers in mind, and I'd like to get to the bottom of this issue to check what needs to be done on all devices so it works with NAT routers, and understand in what context it just can't be used and I should check more NAT-friendly alternatives like IAX. A picture being worth a thousand words, here's the layout I need to use: http://img62.imageshack.us/img62/4077/sipandnatrouters.jpg The PBX server is located in the private LAN behind a NAT router connected to the Internet (I know it'd be easier if it were located in the public network, but this router doesn't support DMZ's so the server has to be in the private network) A couple of (soft|hard)phones are located on the same LAN and connected to the PBX server, along with a PSTN gateway (Linksys 3102 or a Digium PCI card) Remote users using (soft|hard)phones are located somewhere on the Net with dynamic IP's and are also located behind NAT routers I may or may not have control over the local NAT router where the PBX server is located, but I have no control over the remote NAT routers, either because the users don't have the computer knowledge to map ports or because the routers are off-limit (eg. web cafés, hotel LAN's, etc.) Is it possible to configure the PBX server, the (soft|hard)phones, and the PSTN gateway so that the all conversations work fine, no matter the endpoints (POTS caller/local phone, POTS caller/remote phone, local phones, remote phone/local phone)? In which cases may I expect problems, and are there solutions? FWIW, I'm leaning toward using Freeswitch, but I could end up using Asterisk if there are technical advantages to it in this context. Thank you for any info.

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  • How configure 2 Lan cards in Windows 7/8 pc one to connect to Internet and other to Local Network

    - by Maharshi Raval
        I am about to install a dedicated VOIP server in our office. It is a 3CX pbx system on Windows 7/8 machine. The environment currently is a Windows SBS 2011 with 8 client machines. I want to use a dedicated broadband connection for the PBX (3CX) box, but the box also needs to be accessible in the local network as we will be using IP Phones and software IP phones. How configure two network cards on PBX box, so that one will be always used to connect to our SIP host over the Internet and the other will be connected to local network accessible from other client pc to connect to the pbx system. It must be noted that currently the Windows SBS 2011 acts as the Primary Domain Controller and gateway for all the client machines.     I cannot use a load balancer as it will conflict and cause issues within the current setup of our SBS2011 as it is also our Exchange Server. Any input is much appreciated. thanks in advance

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  • How configure 2 Lan cards in Windows 7/8 pc one to connect to Internet and other to Local Network

    - by Maharshi Raval
        I am about to install a dedicated VOIP server in our office. It is a 3CX pbx system on Windows 7/8 machine. The environment currently is a Windows SBS 2011 with 8 client machines. I want to use a dedicated broadband connection for the PBX (3CX) box, but the box also needs to be accessible in the local network as we will be using IP Phones and software IP phones. How configure two network cards on PBX box, so that one will be always used to connect to our SIP host over the Internet and the other will be connected to local network accessible from other client pc to connect to the pbx system. It must be noted that currently the Windows SBS 2011 acts as the Primary Domain Controller and gateway for all the client machines.     I cannot use a load balancer as it will conflict and cause issues within the current setup of our SBS2011 as it is also our Exchange Server. Any input is much appreciated. thanks in advance

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  • How wrong is it to modify the SDP body of a SIP message?

    - by rusbi
    A requirement for the SIP PBX I created for my company was to record all calls passing through it. I solved it by forcing all SIP message to pass through the PBX and to modify the SDP body so the stream passes through it and gets recorded. It works well. I recently found out that this is not allowed. Is there any other way to implement call recording and how "wrong" is this in regard to the protocol?

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  • Asterisk auto Call recording

    - by Manjoor
    We are running asterisk with 8 port FXO. FXO connects to our old PBX (Samsung Office Serv 100). Now we want to record all calls routed through FXO (if it was dialed to outside or comming from outside). Here is the diagram |------|--------------------------------- | |--------------24 Lines ---------- Other clasic Phones PRI------ | PBX |--------------------------------- | | | | | |-----------|---------| | |--8 lines--| |--------- | |-----------|Asterisk |---------- 50 SIP phone |------| | |---------- |---------|---------- Is there a simple way to do this?

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  • Asterisk SIP digest authentication username mismatch

    - by Matt
    I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP image. The 3com phones are communicating SIP with the Asterisk, but are unable to register because they present a digest username value that doesn't match what Asterisk thinks it should. As an example, here are the relevant lines from a successful registration from a soft phone: Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1cac3853" Phone responds: Authorization: Digest username="2321", realm="asterisk", nonce="1cac3853", uri="sip:192.168.254.12", algorithm=md5, response="d32df9ec719817282460e7c2625b6120" For the 3com phone, those same lines look like this (and fails): Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Phone responds: Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" Basically, Asterisk wants to see a username in the Digest username field of 2321, but the 3com phone is sending sip:[email protected]. Anyone know how to tell asterisk to accept this format of username in the digest authentication? Here is the sip.conf info for that extension: [2321] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=1234 qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes mailbox=2321@device host=dynamic dtmfmode=rfc2833 dial=SIP/2321 context=from-internal canreinvite=no callerid=device <2321 allow=ulaw, alaw call-limit=50 ... and for those interested in the grit, here is the debug output of the registration attempt: REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (no NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) confbridge*CLI REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 403 Authentication user name does not match account name Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) Thanks for your input!

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  • Asterisk SIP digest authentication username mismatch

    - by Matt
    I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP image. The 3com phones are communicating SIP with the Asterisk, but are unable to register because they present a digest username value that doesn't match what Asterisk thinks it should. As an example, here are the relevant lines from a successful registration from a soft phone: Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1cac3853" Phone responds: Authorization: Digest username="2321", realm="asterisk", nonce="1cac3853", uri="sip:192.168.254.12", algorithm=md5, response="d32df9ec719817282460e7c2625b6120" For the 3com phone, those same lines look like this (and fails): Server sends: WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Phone responds: Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" Basically, Asterisk wants to see a username in the Digest username field of 2321, but the 3com phone is sending sip:[email protected]. Anyone know how to tell asterisk to accept this format of username in the digest authentication? Here is the sip.conf info for that extension: [2321] deny=0.0.0.0/0.0.0.0 disallow=all type=friend secret=1234 qualify=yes port=5060 permit=0.0.0.0/0.0.0.0 nat=yes mailbox=2321@device host=dynamic dtmfmode=rfc2833 dial=SIP/2321 context=from-internal canreinvite=no callerid=device <2321 allow=ulaw, alaw call-limit=50 ... and for those interested in the grit, here is the debug output of the registration attempt: REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (no NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18580 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c915c33" Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) confbridge*CLI REGISTER sip:192.168.254.12 SIP/2.0 v: SIP/2.0/UDP 192.168.254.157:5060 t: f: i: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER Max-Forwards: 70 m: ;dt=544 Expires: 3600 User-Agent: 3Com-SIP-Phone/V8.0.1.3 Authorization: Digest username="sip:[email protected]", realm="asterisk", nonce="6c915c33", uri="sip:192.168.254.12", opaque="", algorithm=MD5, response="a89df25f19e4b4598595f919dac9db81" X-3Com-PhoneInfo: firstRegistration=no; primaryCallP=192.168.254.12; secondaryCallP=0.0.0.0; --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.254.157 : 5060 (NAT) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 SIP/2.0 403 Authentication user name does not match account name Via: SIP/2.0/UDP 192.168.254.157:5060;received=192.168.254.157 From: To: ;tag=as3fb867e2 Call-ID: fa4451d8-01d6-1cc2-13e4-00e0bb33beb9 CSeq: 18581 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Scheduling destruction of SIP dialog 'fa4451d8-01d6-1cc2-13e4-00e0bb33beb9' in 32000 ms (Method: REGISTER) Thanks for your input!

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  • Asterisk server firewall script allows 2-way audio from incoming calls, but not on outgoing?

    - by cappie
    I'm running an Asterisk PBX on a virtual machine directly connected to the Internet and I really want to prevent script kiddies, l33t h4x0rz and actual hackers access to my server. The basic way I protect my calling-bill now is by using 32 character passwords, but I would much rather have a way to protect The firewall script I'm currently using is stated below, however, without the established connection firewall rule (mentioned rule #1), I cannot receive incoming audio from the target during outgoing calls: #!/bin/bash # first, clean up! iptables -F iptables -X iptables -t nat -F iptables -t nat -X iptables -t mangle -F iptables -t mangle -X iptables -P INPUT ACCEPT iptables -P FORWARD DROP # we're not a router iptables -P OUTPUT ACCEPT # don't allow invalid connections iptables -A INPUT -m state --state INVALID -j DROP # always allow connections that are already set up (MENTIONED RULE #1) iptables -A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT # always accept ICMP iptables -A INPUT -p icmp -j ACCEPT # always accept traffic on these ports #iptables -A INPUT -p tcp --dport 80 -j ACCEPT iptables -A INPUT -p tcp --dport 22 -j ACCEPT # always allow DNS traffic iptables -A INPUT -p udp --sport 53 -j ACCEPT iptables -A OUTPUT -p udp --dport 53 -j ACCEPT # allow return traffic to the PBX iptables -A INPUT -p udp -m udp --dport 50000:65536 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT iptables -A INPUT -p udp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -p tcp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -m multiport -p udp --dports 10000:20000 iptables -A INPUT -m multiport -p tcp --dports 10000:20000 # IP addresses of the office iptables -A INPUT -s 95.XXX.XXX.XXX/32 -j ACCEPT # accept everything from the trunk IP's iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT # accept everything on localhost iptables -A INPUT -i lo -j ACCEPT # accept all outgoing traffic iptables -A OUTPUT -j ACCEPT # DROP everything else #iptables -A INPUT -j DROP I would like to know what firewall rule I'm missing for this all to work.. There is so little documentation on which ports (incoming and outgoing) asterisk actually needs.. (return ports included). Are there any firewall/iptables specialists here that see major problems with this firewall script? It's so frustrating not being able to find a simple firewall solution that enabled me to have a PBX running somewhere on the Internet which is firewalled in such a way that it can ONLY allows connections from and to the office, the DNS servers and the trunk(s) (and only support SSH (port 22) and ICMP traffic for the outside world). Hopefully, using this question, we can solve this problem once and for all.

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  • Convert local time (10 digit number) to a readable datetime format

    - by djerry
    Hey all, I'm working with pbx for voip calls. One aspect of pbx is that you can choose to receive CDR packages. Those packages have 2 timestamps : "utc" and "local", but both seem to always be the same. Here's an example of a timestamp : "1268927156". At first sight, there seems to be no logic in it. So i tried converting it several ways, but with no good result. That value should provide a time around 11am (+1GMT) today. Things i tried: Datetime dt = new Datetime(number); Timespan ts = new Timespan(number); DateTime utc = new DateTime(number + 504911232000000000, DateTimeKind.Utc) and some others i can't remember right now. Am i missing something stupid here? Thanks in advance

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  • Sondage : les bureaux virtuels séduisent de plus en plus les PME pour leur flexibilité et la réduction des coûts qu'ils permettent

    Sondage : les bureaux virtuels séduisent de plus en plus les PME En tête des motivations, flexibilité et réduction des coûts Virtual PBX, fournisseur de solutions virtuelles, vient de publier les résultats d'un sondage sur l'utilisation des bureaux virtuels, désormais au coeur de la politique IT de nombreuses PME. En tête des raisons évoquées par les PDG, propriétaires, partenaires et directeurs de ventes interrogés, arrive le souci de flexibilité (61 %) suivi de près par la « réduction des coûts » avec 54 %. Parmi ces décideurs, 43 % d'entre eux affirment économiser 1.000 dollars ou plus chaque mois sur la location, les équipements IT, la téléphonie et les fournitu...

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  • Fonality: Goodbye Open Source, Hello Cloud

    <b>The VAR Guy:</b> "The company previously positioned itself as an open source IP PBX phone system provider. But going forward, Fonality is pitching itself as a leading provider of cloud-based phone systems and unified communications for small business."

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  • Finding cause of TCP retransmission within a LAN

    - by Surreal
    Hello denizens of Server Fault I have an irritating problem with a LAN of about 100 computers, 2 Windows domain servers, and 12 VoIP phones. Since their installation around a year ago, every week or so, we notice a VoIP phone resetting itself - occasionally in the middle of a call. Simultaneously there are often signs of temporary loss of connection on computers: freezes in explorer while accessing network shares, errors in our administration software due to loss of connection to the database server. I have been doing some Wireshark monitoring on the connection between the VoIP PBX and the rest of the network. Wireshark picks up a clump of retransmitted TCP packets at the times when we record phone restarts. The Wireshark log shows about 2 clusters of retransmissions a day ranging from 5 packets to hundreds. Those in each cluster are mainly between the PBX and some set of the VoIP phones, but not always the same set. Often retransmissions at the same time are to phones connected to the same switch, but sometimes retransmissions occur together to phones at opposite ends of the network. There are usually some coincident retransmissions in passing TCP traffic, for example between client machines and the file servers. The spikes in retransmissions and phone resets do not correlate well with when the network is heavily loaded. They seem to occur slightly more during the day, but most in the evening, when traffic should be decreasing. They occur reasonably often late at night when most computers are turned off and traffic should be lowest. Do you have any ideas that might help diagnose the cause of problems like this? One thing I have not yet tried, but should have, is updating the firmware of all the switches.

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  • Inbound SIP calls through Cisco 881 NAT hang up after a few seconds

    - by MasterRoot24
    I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's simply acting as a modem with one of it's LAN ports connected to FE4 WAN on my Cisco 881. The Cisco 881 get's a DHCP provided IP from my ISP. My LAN is part of default Vlan 1 (192.168.1.0/24). General internet connectivity is working great, I've managed to setup static NAT rules for my HTTP/HTTPS/SMTP/etc. services which are running on my LAN. I don't know whether it's worth mentioning that I've opted to use NVI NAT (ip nat enable as opposed to the traditional ip nat outside/ip nat inside) setup. My reason for this is that NVI allows NAT loopback from my LAN to the WAN IP and back in to the necessary server on the LAN. I run an Asterisk 1.8 PBX on my LAN, which connects to a SIP provider on the internet. Both inbound and outbound calls through the old setup (WAG320N providing routing/NAT) worked fine. However, since moving to the Cisco 881, inbound calls drop after around 10 seconds, whereas outbound calls work fine. The following message is logged on my Asterisk PBX: [Dec 9 15:27:45] WARNING[27734]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission [email protected] for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6528ms with no response [Dec 9 15:27:45] WARNING[27734]: chan_sip.c:3670 retrans_pkt: Hanging up call [email protected] - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). (I know that this is quite a common issue - I've spend the best part of 2 days solid on this, trawling Google.) I've done as I am told and checked https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions. Referring to the section "Other SIP requests" in the page linked above, I believe that the hangup to be caused by the ACK from my SIP provider not being passed back through NAT to Asterisk on my PBX. I tried to ascertain this by dumping the packets on my WAN interface on the 881. I managed to obtain a PCAP dump of packets in/out of my WAN interface. Here's an example of an ACK being reveived by the router from my provider: 689 21.219999 193.x.x.x 188.x.x.x SIP 502 Request: ACK sip:[email protected] | However a SIP trace on the Asterisk server show's that there are no ACK's received in response to the 200 OK from my PBX: http://pastebin.com/wwHpLPPz In the past, I have been strongly advised to disable any sort of SIP ALGs on routers and/or firewalls and the many posts regarding this issue on the internet seem to support this. However, I believe on Cisco IOS, the config command to disable SIP ALG is no ip nat service sip udp port 5060 however, this doesn't appear to help the situation. To confirm that config setting is set: Router1#show running-config | include sip no ip nat service sip udp port 5060 Another interesting twist: for a short period of time, I tried another provider. Luckily, my trial account with them is still available, so I reverted my Asterisk config back to the revision before I integrated with my current provider. I then dialled in to the DDI associated with the trial trunk and the call didn't get hung up and I didn't get the error above! To me, this points at the provider, however I know, like all providers do, will say "There's no issues with our SIP proxies - it's your firewall." I'm tempted to agree with this, as this issue was not apparent with the old WAG320N router when it was doing the NAT'ing. I'm sure you'll want to see my running-config too: ! ! Last configuration change at 15:55:07 UTC Sun Dec 9 2012 by xxx version 15.2 no service pad service tcp-keepalives-in service tcp-keepalives-out service timestamps debug datetime msec localtime show-timezone service timestamps log datetime msec localtime show-timezone no service password-encryption service sequence-numbers ! hostname Router1 ! boot-start-marker boot-end-marker ! ! security authentication failure rate 10 log security passwords min-length 6 logging buffered 4096 logging console critical enable secret 4 xxx ! aaa new-model ! ! aaa authentication login local_auth local ! ! ! ! ! aaa session-id common ! memory-size iomem 10 ! crypto pki trustpoint TP-self-signed-xxx enrollment selfsigned subject-name cn=IOS-Self-Signed-Certificate-xxx revocation-check none rsakeypair TP-self-signed-xxx ! ! crypto pki certificate chain TP-self-signed-xxx certificate self-signed 01 quit no ip source-route no ip gratuitous-arps ip auth-proxy max-login-attempts 5 ip admission max-login-attempts 5 ! ! ! ! ! no ip bootp server ip domain name dmz.merlin.local ip domain list dmz.merlin.local ip domain list merlin.local ip name-server x.x.x.x ip inspect audit-trail ip inspect udp idle-time 1800 ip inspect dns-timeout 7 ip inspect tcp idle-time 14400 ip inspect name autosec_inspect ftp timeout 3600 ip inspect name autosec_inspect http timeout 3600 ip inspect name autosec_inspect rcmd timeout 3600 ip inspect name autosec_inspect realaudio timeout 3600 ip inspect name autosec_inspect smtp timeout 3600 ip inspect name autosec_inspect tftp timeout 30 ip inspect name autosec_inspect udp timeout 15 ip inspect name autosec_inspect tcp timeout 3600 ip cef login block-for 3 attempts 3 within 3 no ipv6 cef ! ! multilink bundle-name authenticated license udi pid CISCO881-SEC-K9 sn ! ! username xxx privilege 15 secret 4 xxx username xxx secret 4 xxx ! ! ! ! ! ip ssh time-out 60 ! ! ! ! ! ! ! ! ! interface FastEthernet0 no ip address ! interface FastEthernet1 no ip address ! interface FastEthernet2 no ip address ! interface FastEthernet3 switchport access vlan 2 no ip address ! interface FastEthernet4 ip address dhcp no ip redirects no ip unreachables no ip proxy-arp ip nat enable duplex auto speed auto ! interface Vlan1 ip address 192.168.1.1 255.255.255.0 no ip redirects no ip unreachables no ip proxy-arp ip nat enable ! interface Vlan2 ip address 192.168.0.2 255.255.255.0 ! ip forward-protocol nd ip http server ip http access-class 1 ip http authentication local ip http secure-server ip http timeout-policy idle 60 life 86400 requests 10000 ! ! no ip nat service sip udp port 5060 ip nat source list 1 interface FastEthernet4 overload ip nat source static tcp x.x.x.x 80 interface FastEthernet4 80 ip nat source static tcp x.x.x.x 443 interface FastEthernet4 443 ip nat source static tcp x.x.x.x 25 interface FastEthernet4 25 ip nat source static tcp x.x.x.x 587 interface FastEthernet4 587 ip nat source static tcp x.x.x.x 143 interface FastEthernet4 143 ip nat source static tcp x.x.x.x 993 interface FastEthernet4 993 ip nat source static tcp x.x.x.x 1723 interface FastEthernet4 1723 ! ! logging trap debugging logging facility local2 access-list 1 permit 192.168.1.0 0.0.0.255 access-list 1 permit 192.168.0.0 0.0.0.255 no cdp run ! ! ! ! control-plane ! ! banner motd Authorized Access only ! line con 0 login authentication local_auth length 0 transport output all line aux 0 exec-timeout 15 0 login authentication local_auth transport output all line vty 0 1 access-class 1 in logging synchronous login authentication local_auth length 0 transport preferred none transport input telnet transport output all line vty 2 4 access-class 1 in login authentication local_auth length 0 transport input ssh transport output all ! ! end ...and, if it's of any use, here's my Asterisk SIP config: [general] context=default ; Default context for calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) ; Optionally add a port number, 192.168.1.1:5062 (default is port 5060) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet ; Specifying a port in a SIP peer definition or ; when dialing outbound calls will supress SRV ; lookups for that peer or call. directmedia=no ; Don't allow direct RTP media between extensions (doesn't work through NAT) externhost=<MY DYNDNS HOSTNAME> ; Our external hostname to resolve to IP and be used in NAT'ed packets localnet=192.168.1.0/24 ; Define our local network so we know which packets need NAT'ing qualify=yes ; Qualify peers by default dtmfmode=rfc2833 ; Set the default DTMF mode disallow=all ; Disallow all codecs by default allow=ulaw ; Allow G.711 u-law allow=alaw ; Allow G.711 a-law ; ---------------------- ; SIP Trunk Registration ; ---------------------- ; Orbtalk register => <MY SIP PROVIDER USER NAME>:[email protected]/<MY DDI> ; Main Orbtalk number ; ---------- ; Trunks ; ---------- [orbtalk] ; Main Orbtalk trunk type=peer insecure=invite host=sipgw3.orbtalk.co.uk nat=yes username=<MY SIP PROVIDER USER NAME> defaultuser=<MY SIP PROVIDER USER NAME> fromuser=<MY SIP PROVIDER USER NAME> secret=xxx context=inbound I really don't know where to go with this. If anyone can help me find out why these calls are being dropped off, I'd be grateful if you could chime in! Please let me know if any further info is required.

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  • Finding cause of TCP retransmission within a LAN

    - by Surreal
    Hello denizens of Server Fault I have an irritating problem with a LAN of about 100 computers, 2 Windows domain servers, and 12 VoIP phones. Since their installation around a year ago, every week or so, we notice a VoIP phone resetting itself - occasionally in the middle of a call. Simultaneously there are often signs of temporary loss of connection on computers: freezes in explorer while accessing network shares, errors in our administration software due to loss of connection to the database server. I have been doing some Wireshark monitoring on the connection between the VoIP PBX and the rest of the network. Wireshark picks up a clump of retransmitted TCP packets at the times when we record phone restarts. The Wireshark log shows about 2 clusters of retransmissions a day ranging from 5 packets to hundreds. Those in each cluster are mainly between the PBX and some set of the VoIP phones, but not always the same set. Often retransmissions at the same time are to phones connected to the same switch, but sometimes retransmissions occur together to phones at opposite ends of the network. There are usually some coincident retransmissions in passing TCP traffic, for example between client machines and the file servers. The spikes in retransmissions and phone resets do not correlate well with when the network is heavily loaded. They seem to occur slightly more during the day, but most in the evening, when traffic should be decreasing. They occur reasonably often late at night when most computers are turned off and traffic should be lowest. Do you have any ideas that might help diagnose the cause of problems like this? One thing I have not yet tried, but should have, is updating the firmware of all the switches.

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  • Asterisk relay between multiple subnets

    - by immoune
    I wonder what's the best way to go when you have phones on multiple networks which are not directly reachable. I have 3 networks 10.3.x.x 10.6.x.x 10.17.x.x My asterisk server resides on the 10.3.0.5 IP. The machines from the 10.6 and 10.17 networks are routed here through VPN tunnels. At this point we don't talk about NAT anywhere on the network just pure routing. Since the 10.3.0.5 PBX has routes back to all the subnet's it has no problem to communicate with softphones/hardphones from these ranges. The problem comes from that Asterisk (as far as I understand) only responsible for the SIP communication part not the Audio/Video transmission which is in P2P fashion done between the devices. So although a client using sipdroid from 10.6.x.x is able to connect to the pbx (10.3.0.5) and dial a bria client on the 10.17.x.x network once the phone rings out and the call establishes no audio will be transmitted simply because it has no way to directly connect there. For this there are multiple solutions described in this text: http://msdn.microsoft.com/en-us/library/ee480411%28v=winembedded.60%29.aspx What I would prefer is to keep these networks segregated as they are now. What would be the best solution? Is it possible to actually relay through all the audio/video information through the Asterisk server? That would be the best in my case, I using Astlinux there which has a lot of other parts. Thanks

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  • Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

    - by slashp
    I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic (from Lync) & UDP traffic (to the trixbox, as Asterisk 1.4 does not support SIP over TCP). Our Lync box IP: 10.100.10.41 Our Kamailio box IP: 10.100.10.44 Our trixbox IP: 10.100.10.2 The issue I'm running into is as follows when enabling SIP debugging for the Kamailio box: <--- SIP read from 10.100.10.44:5060 ---> PRACK sip:TNECLTSLY01.contoso.com:5068;transport=Tcp;maddr=10.100.10.41 SIP/2.0 FROM: <sip:9121;[email protected];user=phone>;epid=CF2380792B;tag=4852bab430 TO: <sip:[email protected];user=phone>;epid=CF2380792B;tag=3684a6a24e CSEQ: 24 PRACK CALL-ID: 192daae6-00e1-4140-bddd-0394b35d475b MAX-FORWARDS: 70 Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d VIA: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK159fc989 CONTACT: <sip:TNECLTSLY01.contoso.com:5068;transport=Tcp;maddr=10.100.10.41> CONTENT-LENGTH: 0 USER-AGENT: RTCC/4.0.0.0 MediationServer RAck: 1 23 INVITE <-------------> --- (12 headers 0 lines) --- Sending to 10.100.10.44 : 5060 (NAT) <--- Transmitting (NAT) to 10.100.10.44:5060 ---> SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d;received=10.100.10.44 Via: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK159fc989 From: <sip:9121;[email protected];user=phone>;epid=CF2380792B;tag=4852bab430 To: <sip:[email protected];user=phone>;epid=CF2380792B;tag=3684a6a24e Call-ID: 192daae6-00e1-4140-bddd-0394b35d475b CSeq: 24 PRACK User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> trixbox1*CLI> <--- SIP read from 10.100.10.44:5060 ---> ACK sip:[email protected];user=phone SIP/2.0 FROM: "John Jones"<sip:9121;[email protected];user=phone>;tag=4852bab430;epid=CF2380792B TO: <sip:[email protected];user=phone>;tag=3684a6a24e;epid=CF2380792B CSEQ: 23 ACK CALL-ID: 192daae6-00e1-4140-bddd-0394b35d475b MAX-FORWARDS: 70 Via: SIP/2.0/UDP 10.100.10.44;branch=z9hG4bKcydzigwkX;i=d VIA: SIP/2.0/TCP 10.100.10.41:51677;branch=z9hG4bK79a21c CONTENT-LENGTH: 0 My SIP trunk on the trixbox looks like this: [from-lync] exten => _+4XXX!,1,Noop(Stripping + from start of number) exten => _+4XXX!,n,Goto(from-internal,${EXTEN:1}) Though I am still having no luck getting the + stripped or the call to go through. Any ideas would be greatly appreciated. Thank you! -slashp

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  • Can a FreePBX backup be restored to a different version?

    - by Tim Long
    I run a small PBX based on the FreePBX distro of Asterisk. The installation has been steadily upgraded but for various reasons, we want to start again on a new server with a clean install from the distribution media. Will I be able to take a backup from the old server and restore it to the new server, even though the installs are different versions? How sensitive are FreePBX backups to the build version? Is it possible to get at least a partial restore?

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  • Vlan on astaro 120

    - by Crash893
    (I'm not 100% sure where networking/router questions go this is my best guess) I have a astaro (sophos) white UTM 120 router for work I also have about 11 Voip phones with an externaly hosted pbx (company name = pingtone) Is there any advantage to setting up the phones on a vlan vs making a qos rule that all traffic to my tftp server gets right of way? networking is still a little soft to me Thanks

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  • Lync Server 2010

    - by ManojDhobale
    Microsoft Lync Server 2010 communications software and its client software, such as Microsoft Lync 2010, enable your users to connect in new ways and to stay connected, regardless of their physical location. Lync 2010 and Lync Server 2010 bring together the different ways that people communicate in a single client interface, are deployed as a unified platform, and are administered through a single management infrastructure. Workload Description IM and presence Instant messaging (IM) and presence help your users find and communicate with one another efficiently and effectively. IM provides an instant messaging platform with conversation history, and supports public IM connectivity with users of public IM networks such as MSN/Windows Live, Yahoo!, and AOL. Presence establishes and displays a user’s personal availability and willingness to communicate through the use of common states such as Available or Busy. This rich presence information enables other users to immediately make effective communication choices. Conferencing Lync Server includes support for IM conferencing, audio conferencing, web conferencing, video conferencing, and application sharing, for both scheduled and impromptu meetings. All these meeting types are supported with a single client. Lync Server also supports dial-in conferencing so that users of public switched telephone network (PSTN) phones can participate in the audio portion of conferences. Conferences can seamlessly change and grow in real time. For example, a single conference can start as just instant messages between a few users, and escalate to an audio conference with desktop sharing and a larger audience instantly, easily, and without interrupting the conversation flow. Enterprise Voice Enterprise Voice is the Voice over Internet Protocol (VoIP) offering in Lync Server 2010. It delivers a voice option to enhance or replace traditional private branch exchange (PBX) systems. In addition to the complete telephony capabilities of an IP PBX, Enterprise Voice is integrated with rich presence, IM, collaboration, and meetings. Features such as call answer, hold, resume, transfer, forward and divert are supported directly, while personalized speed dialing keys are replaced by Contacts lists, and automatic intercom is replaced with IM. Enterprise Voice supports high availability through call admission control (CAC), branch office survivability, and extended options for data resiliency. Support for remote users You can provide full Lync Server functionality for users who are currently outside your organization’s firewalls by deploying servers called Edge Servers to provide a connection for these remote users. These remote users can connect to conferences by using a personal computer with Lync 2010 installed, the phone, or a web interface. Deploying Edge Servers also enables you to federate with partner or vendor organizations. A federated relationship enables your users to put federated users on their Contacts lists, exchange presence information and instant messages with these users, and invite them to audio calls, video calls, and conferences. Integration with other products Lync Server integrates with several other products to provide additional benefits to your users and administrators. Meeting tools are integrated into Outlook 2010 to enable organizers to schedule a meeting or start an impromptu conference with a single click and make it just as easy for attendees to join. Presence information is integrated into Outlook 2010 and SharePoint 2010. Exchange Unified Messaging (UM) provides several integration features. Users can see if they have new voice mail within Lync 2010. They can click a play button in the Outlook message to hear the audio voice mail, or view a transcription of the voice mail in the notification message. Simple deployment To help you plan and deploy your servers and clients, Lync Server provides the Microsoft Lync Server 2010, Planning Tool and the Topology Builder. Lync Server 2010, Planning Tool is a wizard that interactively asks you a series of questions about your organization, the Lync Server features you want to enable, and your capacity planning needs. Then, it creates a recommended deployment topology based on your answers, and produces several forms of output to aid your planning and installation. Topology Builder is an installation component of Lync Server 2010. You use Topology Builder to create, adjust and publish your planned topology. It also validates your topology before you begin server installations. When you install Lync Server on individual servers, the installation program deploys the server as directed in the topology. Simple management After you deploy Lync Server, it offers the following powerful and streamlined management tools: Active Directory for its user information, which eliminates the need for separate user and policy databases. Microsoft Lync Server 2010 Control Panel, a new web-based graphical user interface for administrators. With this web-based UI, Lync Server administrators can manage their systems from anywhere on the corporate network, without needing specialized management software installed on their computers. Lync Server Management Shell command-line management tool, which is based on the Windows PowerShell command-line interface. It provides a rich command set for administration of all aspects of the product, and enables Lync Server administrators to automate repetitive tasks using a familiar tool. While the IM and presence features are automatically installed in every Lync Server deployment, you can choose whether to deploy conferencing, Enterprise Voice, and remote user access, to tailor your deployment to your organization’s needs.

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