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Ganesh

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Everything posted by Ganesh

  1. We have a setup running in which there is one way speech. For inbound calls via the trunk the far end person can hear the agent but the agent cannot hear any voice. Below is log. The firewall is open for UDP ports. Can you see if we need to do any configuration changes in Pbxnsip? [9] 2009/07/24 14:46:25: SIP Rx udp:192.168.200.204:5060: OPTIONS sip:201@192.168.200.204 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.204;branch=z9hG4bKac608188397 Max-Forwards: 70 From: <sip:201@192.168.200.204>;tag=1c608184426 To: <sip:201@192.168.200.204> Call-ID: 60818406411200020958@192.168.200.2
  2. We upgraded the CS410 (white box) to the latest version of pbxnsip application. The upgrade was done as there was no CLI displayed for incoming calls on FXO port. After upgrade, the FXO ports are not working. There is no incoming and out going calls on FXO ports. While the PSTN line is connected to FXO port, the LED is not flashing when there is an incoming call. No logs are captured for an incoming call on FXO port. Below is the logs captured while an out going call is made. In these logs it shows INVITE, TRYING, RINGING, 200 OK etc. But the call does not go thru. The dialed PSTN number d
  3. Does pbxnsip support H.281 for far end camera control? Is there a way we can do this on SIP? The customer is using standard video phones with pbxnsip. Regards Ganesh
  4. Hi, We are running PBXNSIP version 3.0 on Linux centOS. The system was working fine and no changes were made. Its suddenly down and we have restarted it number of times but the service is not starting. [root@cel-sip ~]# service httpd status httpd (pid 2390 2388 2387 2386 2385 2384 2383 2382 2294) is running.. [root@cel-sip ~]# cd /etc/init.d [root@cel-sip init.d]# ./pbxnsip restart Stopping PBX:FAILED] Starting PBX: [root@cel-sip init.d]# ./pbxnsip status pbxctrl is stopped [root@cel-sip init.d]# [root@cel-sip ~]# more /etc/init.d/pbxnsip #!/bi
  5. We configured the outbound proxy and specified to use TLS (as Avaya only supports TLS on direct SIP trunk without Avaya SES). We were still unable to route calls between the two systems. Below is the logs captured. 59999 is the Avaya phone extension. This link (http://www.avayausers.com/showthread.php?t=10700) says that "TCP Sip is supported, but UDP SIP is not supported without the Avaya Sip Server. (SES) Even then you still need a session border controller" (we are not using the Avaya SES here). It also says SBC is required while using the SIP trunk on Avaya. Since we are integra
  6. Sometimes the registration of phones goes off. If the pbxnsip service is restarted, the phones then gets registered. Below is the logs taken before the service was restarted. [7] 2008/07/14 12:06:09: SIP Rx udp:10.255.109.71:5060: REGISTER sip:10.255.10.41:5060 SIP/2.0 Via: SIP/2.0/UDP 10.255.109.71:5060;branch=z9hG4bK8277290331884624686;rport From: 9 <sip:9@10.255.10.41:5060>;tag=72856 To: 9 <sip:9@10.255.10.41:5060> Call-ID: 156118939-2437221063@10.255.109.71 CSeq: 79 REGISTER Contact: <sip:9@10.255.109.71:5060> Max-Forwards: 70 Expires: 60 User-Agent: Vo
  7. We are integrating pbxnsip with Avaya system on SIP trunk. We have configured SIP trunk (gateway mode) on pbxnsip and Avaya. But unable to route calls both ways between the two system. Attached is the logs and wireshark traces captured. 4229 and 4431 is extension (Avaya phones) in Avaya. 2201 and 2202 is extensions (Snom phones) on pbxnsip. Avaya uses port 5061 and supports only TLS on SIP trunk. Please let me know if you can get some information from these logs. Below is one more log taken from pbxnsip while trying to call Avaya phones. [5] 2008/07/07 04:23:17: SIP port accept fr
  8. Customer is having MySQL 5 database for their office employees. They are planning to integrate the PBXNSIP IVR tree with this. The application is if an employee calls from outside to the office, the IVR tree of PBXNSIP directs the call to the database server. The employee then needs to enter his ID number and password for authentication. He will then get a series of options from the database IVR. For ex: If the employee has to reach the HR department and apply for a leave he will then enter the date for leave application. The HR department will then get an alert from the database server. T
  9. Can we have a feature code and password to block the outgoing calls of the user? Each user will dial the feature code and enter the password to lock/unlock the phone.
  10. Does pbxnsip support G 723.1 codec?
  11. Hi Can you please tell me what is the BHCC and BHCA values currently available in PBXnSIP? Our customer is using a dual core P4 server with 4 GB RAM. Regards Ganesh
  12. How do we change the pbxnsip logo in Linux redhat and Suse? Can someone help?
  13. We are installing pbxnsip in Linux Suse 9.2. I downloaded the file pbxctrl-suse10-2.1.6.2448. I checked the wiki on how to install in Linux but as we are not very familiar with Linux its all looking like greek to us. I appreciate if someone could send me few steps on how to install pbxnsip in Suse. Maybe a step by step procedure after downloading and how to install will be really helpful. The downloaded pbx file is for Suse 10 but we are using Suse 9.2. Is that ok or should we switch to Suse 10.x? Regards Ganesh
  14. I created an IVR and added the default MoH file of PBXNSIP. From a registered user if i dial the IVR number the music is played continuously. I configured a trunk (SIP gateway) in PBXNSIP and specified the incoming calls to land on IVR. From the FXS gateway (not registered to PBXNSIP) I was able to send the MOH request to PBXNSIP but the music is not played to the analog extensions. In the logs i see the request from the FXS gateway is forwarded to the IVR. Why is it not playing the music? Is there any other settings? The logs are attached. MOH_logs.txt
  15. Yes. I will use this account to send MOH request from the gateway to play music when the analog phones puts a call on hold.
  16. I want to use the MOH of PBXnSIP to be played to the analog phones connected to a FXS gateway. The FXS gateway as a provision to use external MOH server for playing the music. It is not registered or configured as trunk with the PBXnSIP. In the gateway i specified the MOH URL as "moh.wav@IP address of pbxnsip" and see the gateway sending the request to PBXnSIP but says "not found". How do i create a MOH account in the PBXnSIP? Is this possible? Can i create an extension and point it to play a MOH for the gateway?
  17. Eyebeam is using G.729 only and also on the PBXNSIP we have set G.729 as the first codec in settings and in trunk. How do we see if transcoding is taking place? When eyebeam is directly configured with the ITSP details it is using G.729 and the voice is clear.
  18. We have version 2.1.2.2292 (Win32) which is installed in the LAN. The domain is created as "localhost". The LAN IP is natted to a public IP. Phones within the LAN are using the natted IP as registrar and getting registered to the PBX. The phones outside the office are using this public IP to register. The phones at some office are registering with the PBX. But in the PBX trace for registered phones, we see Via: SIP/2.0/UDP 0.0.0.0:1040 and Contact: <sip:109@0.0.0.0:1040>. Is this normal? Why is it not displaying the IP address? Registered trace is given below. But at some office
  19. We have configured an ITSP trunk with version 2.1.2.2292. We use the eye beam soft phone and when we make an outbound call to a mobile number, the voice is not clear at both the end. The sound level varies and sometimes goes silent in between. If we configure the same ITSP details directly on the eye beam soft phone and make calls, the voice is clear on both the side. Below is the logs captured from pbxnsip after making a call. INVITE sip:0791394772@196.22.138.50;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK-1408f7619cd8548827d78e33ae5c154f;rport From: "saar" &
  20. Hi We have a customer who is looking for integrating the address book of microsoft outlook with PBXNSIP such that they use the microsoft outlook to dial the call. They click on the address book and which then sends a dial out to PBXNSIP. I belive another vendor as told the customer this is supported by their application. Is this possible with PBXNSIP? Regards Ganesh
  21. We are using PBXNSIP version 2.1.2.2292 (Win32). On the same server we also have a Xlite soft phone registered to pbxnsip. If we use the Xlite soft phone and make any calls to the registered users in the domain (other soft phone/hard phone) we have one way voice. The Xlite on the server can hear the other person whereas the far end phone can hear nothing. We also have an ITSP trunk registered. If the Xlite on server dials a call via the trunk then the same problem is observed. But the other registered users can dial out via the trunk with both side voice. Also we tried installing many
  22. Our customer is using pbxnsip version 2.1.2.2292 (Win32). We added an ITSP trunk of a service provider. The trunk is not registering with pbxnsip. We see an error "408 Request Timeout" or "500 Address Resolution Failed". Instead of the ITSP domain name (sip99.telfreesa.com) we tried using its associated IP address in the "Trunk" "domain" settings but the trunk still does not register. Looks like this ITSP trunk (sip99.telfreesa.com) can only be registered using its domain name and not its IP address. On the same pbxnsip server we have a X lite soft phone installed and if the X lite soft p
  23. We have installed PBXNSIP version 2.1.1 2201 (windows 32) on Windows Vista. When we make a call from one extension to other, there is no voice between them. The default firewall is disabled and no other application is installed in the system. If we dial a IVR number or conference, we can hear the recorded message from the system but not the voice between the users. The same version is working fine if we replace the server with Windows XP machine. Is there any others settings in Vista to be changed? Regards Ganesh
  24. This customer has the existing database in mysql and language of the intranet site is jsp. The fields shown in the screen correspond to the following table.field: Company = aa_hierachy.countryCode Department = aa_hierachy.deptCode Section = aa_hierachy.sectionCode Staff Number = aa_hierachy.staffNumber Name = security_user.firstName Office Direct Line = aa_hierachy.contactOfficeDirectLineNumber Office Ext = aa_hierachy.contactOfficeGeneralNumber Handphone = aa_hierachy.contactHpNumber Email = security_user.email1 If we use TSP can get something like this with a "click
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