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Questions tagged [sip]

The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions. Sessions may consist of one or several media streams.

The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol (IP). The protocol can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions. Sessions may consist of one or several media streams.

The SIP protocol is an Application Layer protocol designed to be independent of the underlying Transport Layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP).It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).

Source: wikipedia

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Avaya 9630 for SIP

Is there any way to make the 9630 Desktop Headset work just as a SIP IP telephone, without the extensive need for additional Avaya equipment. All of our infrastructure is Cisco and we may receive about 80 Avaya deivces and I am wondering if they…
xciter
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Lync Server 2010 with Hosted VoIP PBX

We just deployed Lync Server 2010 in our organization and it is working great so far. The next step for us is to enable enterprise voice so that we can replace our telephones with service that is handled 100% by Lync. This is where I am at a…
kmehta
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Asterisk Incoming Calls Don't Route to Extensions

We have just setup Asterisk 1.6 with FreePBX 2.8. We configured SIP Trunks with our Polish SIP Provider (a total of 3 trunks) in the system. Using a softphone registered to the only extension in the system we are able to make calls out. However,…
Brent Pabst
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SIP configuration with a Sonicwall firewall

I am working with a partner to enable SIP between a local SIP server and 2 Polycom phones located across a WAN connection. The server is located behind a Cicso ASA with SIP translation enabled, and the on the partner's side there is a Sonicwall…
chills42
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Test-service on Internet for testing incoming INVITE

I am trying to set up Asterisk at home. I think I'm having trouble configuring my firewall, so that inbound traffic is accepted, but I am not sure. I got the idea that, perhaps, there is a service out on the Internet, where I can, though a…
leiflundgren
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problem with red5phone unable to connect

i'm looking at red5 and red5phone to make it works. i am getting an error netconnection: Status: 1 rtmpconnection #phone _onStatus NetConnection.Connect.Failed netconnection: Status: 0 connection failed to rtmp://localhost/sip : connection failed is…
sunil221
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Lost Unified Messaging (Exhcnage 2007) from Call Manager after external SIP Provider

ServerFault, After some really great trouble-shooting sessions with help from some people on Experts-Exchange... I've managed to get my 2621xm connected to BroadVoice's SIP Service through Call Manager Express (v3.1). It works great! But there's…
user43207
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RAS server using SIP/h323?

We currently have a few modems (using standard POTS lines from the telco) connected to a server 2003 box running RRAS for remote access connections (mainly remote data pushes). We also have a Cisco VoIP system in place in our building utilizing two…
Dan
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How do I configure the SIP source interface on a Cisco router so that it sticks?

I manage a cisco router acting as a SIP gateway. In order to get it to register to the SIP provider, the connection needs to come from the right IP address. This is done with the following lines in the switch config: voice service voip sip bind…
Josh
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How different is "Skype Connect" from "Skype for Asterisk"

We already use "Skype for Asterisk" channels to connect Skype to Asterisk PBX. Skype has a new offering called "Skype Connect" (or formerly "Skype for SIP"). What are the differences?
Chirag N.R.
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Asterisk Server with SIP Account Routing to Cell Phones

I'm having trouble finding the exact documentation to do the following: I have a SIP Account. I want my Asterisk Server on a VPS somewhere in the United States to accept the credentials of the SIP Account. When people call into my SIP account…
John
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Firewall issue with multiple SIP PROXY / REGISTRAR servers

We have a pair of Internet-facing SIP PROXY/REGISTRAR servers (for resilienced and load-balancing). When a SIP phone registers, it will be handled by one of the REGISTRAR servers (round-robin DNS) - and since this registration is renewed, the…
MikeBrom
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One way audio on SIP phone on one call direction.. where to look next?

Got 2 SIP gateways (ZultysMX250). Sip trunk between them, they are in separate locations, different subnets 192.168.XX and 192.168.YY connected by IPSec VPN. Some phones work perfectly. But mine... (and I'm the admin.. grrr) is a bit older, and the…
Tom Newton
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T38modem manuals?

Are there any T38modem users here? I'm trying to figure out how to call T38modem with SIP. I've got everything except the --route option for receiving. I know my own phone number, but I am not sure how to set it up. Currently, I have: --route…
Brian Postow
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Setup Kamailio to accept only register request with FQDN and block IP request

I have set up Kamailio and am quite new to the process, I'm able to register with FQDN and IP directly to Kamailio but my main setup is Kamailio as edge proxy to asterisk to allow register only with FQDN and not IP, route { if…
Mohammad Sarshar
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