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Questions tagged [sip-server]

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the network.

The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.

The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions consisting of one or several media streams. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.

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How manual dial in vicidial is implemented with asterisk?

I logged in using agc (agent) panel in vicidial and just to test manual dial, I just executed following MySql query fron cmd. UPDATE vicidial_live_agents set external_dial='12122351880!!YES!NO!YES!!1478530720!!!!!!' where user='1001'; After this ,…
Anup_Tripathi
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Are there alternatives to SIP Trunking?

I'm not sure if this is a stackoverflow question but, To my understanding SIP Trunking is a VOIP protocol that allows a person to call through a phoneline and than some datacenter will convert that into an internet call. My question is, is it the…
Kaasstengel
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Is it possible to forward sip requests via a node js server to another asterisk server?

I am working to make a sip client for calling. As a server my company uses asterisk(VOS3000). The server doesn't support web socket. And now I want to know if there is any way to make a sip client using javascript being in my situation. May be its…
KOUSIK MANDAL
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PJSIP - Pjsua2 android: How to add headers while calling?

Below is my source code for pjsip calling - String buddy_uri = item.get("uri"); SipHeaderVector sipHeaderVector = new SipHeaderVector(2); SipHeader sipHeader1 = new SipHeader(); sipHeader1.setHName("Header1"); …
Ravi D
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How to change the opensips default port number 5060 to some other port?

I tried changing the opensips port number form 5060 to some other port in opensips.cfg file. port=5060 - I changed this to port=6060. Restarted opensips after this. But this change is not working. How to make sure this is working?
Giri
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How can I load test and performance test an asterisk sip server to find out how many concurrent calls a server can handle?

I have deployed asterisk server in AWS. 1) I would like to know how many calls it can handle concurrently at the same time? 2) Is it possible to do a load test with several phone numbers? How to simulate it? Any tools?
sofs1
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Can I have a new real time SIP family name other than the well defined family names in asterisk server?

In this link http://www.voip-info.org/wiki/index.php?page_id=1735 [settings] => ,~np~[~/np~,table_name~np~]~/np~ sippeers => mysql,asterisk,sip_peers sipusers => mysql,asterisk,sip_users queues =>…
user3705456
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Why does 'odbc show' command returns empty ODBC and DSN settings?

I have my odbc.ini and odbcinst.ini as follows odbcinst.ini [asterisk-connector] Description = MySQL connection to 'asterisk' database Driver = MySQL Database = asterisk Server = localhost UID = asterisk password =…
user3705456
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Is there a difference between asterisk -rvvvvvv and /etc/init.d/asterisk restart

is there a difference between asterisk -rvvvvvv and /etc/init.d/asterisk restart Which one will make use of latest changes of sip.conf and extensions.conf? Though I do sip reload and diaplan reload I would like to know the difference between …
user3705456
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Either the outbound or inbound call only work in asterisk setup, not both. Why?

This is my sip.conf ; inbound configuration [nexmo-sip] fromdomain=sip.nexmo.com type=friend context=nexmo insecure=port,invite nat=no ;Add your codec list here. ; Note: Use "ulaw" for US only, "alaw" for the rest of the…
user3705456
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Re-INVITE during call results with 481 Call Does Not Exist

I've got a FreeSwitch server (1.4.26 on Ubuntu). When redirecting an incoming call to an external server, 30 minutes after the call connected, I've getting a RE-INVITE message from the target server. My FreeSwitch server responds with "481 Call Does…
Eliram
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How to intercept / observe SIP traffic through fiddler or wireshark?

I am using a free caller android app and they restrict me with limited credits. I would like to explore more about how this app works. So I started decompiling the APK file and intercepting networking requests using Fiddler. I intercepted requests,…
Varun Kakumani
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Does Skype For Business (Lync 2015) support inbound History-Info headers

Sniffing with Wireshark I can see that an inbound INVITE from our PSTN sip trunk contains a history-info header, but this header is not forwarded to our endpoint application server where I want to pick it up. The sip logs on the mediation server…
MatiasK
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How to terminate a sip call with some fixed time

I have written a service for handling sip call. I want to make some additional feature to restrict a call time by configuring a fix time or handling the call time value by sending with some parameter. Once a sip call got established generally it…
abhisek
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Is it necessary to get authorization for de register also?

I am working with SIP.Currently I am seing a scenario in which a register with Expires header value 0 is going to the server.The server gives a 401 unauthorized and the phone sends a register again .This time also the register goes with the…
achoora
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