Re: [OpenSIPS-Users] memory consumed by t_relay

Hi Bogdan-Andrei, When you get the memory error, could you please do : opensipsctl fifo get_statistics all and send me the output (off list) please? When about 10,000 calls, 10kcalls.log http://opensips-open-sip-server.1449251.n2.nabble.com/file/n7586442/10kcalls.log . When about 20,000

Re: [OpenSIPS-Users] Mediaproxy relay on CentOS 6 VPS

Hi Muhammad, The (edited) results for those commands you suggested running are as follows: processor : 0 vendor_id : AuthenticAMD cpu family : 16 model : 2 model name : Quad-Core AMD Opteron(tm) Processor 2352 stepping: 3 cpu MHz : 1050.000

Re: [OpenSIPS-Users] Registrar not saving received from Path header

Hi Nathaniel, Well, the logs shows that save() does not receive any flags as params...everything indicates that you do not have the params or you are using the wrong config file. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/22/2013 08:22

Re: [OpenSIPS-Users] Do_Routing problem

Kaled, You need to pay a bit of attention on what you are doing and try to understand things - simply throeing some lines into the script, without being consistent, will not make your script work and simply waste our time here. So, one more

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

I think that is retransmission of ACK packet because it didn't get its 200 ok back. Regards, Qasim On Tue, May 21, 2013 at 10:08 PM, Bogdan-Andrei Iancu bog...@opensips.orgwrote: ** Hi Qasim, Looking at the ACK related logs, I see you get the script log Sequencial 'ACK' request from

Re: [OpenSIPS-Users] Registrar not saving received from Path header

Hello Bogdan, I have validated the script and that i am passing a parameter. I also changed the debug log statement that I displayed right before the save() and I still get the same output. Here is the code that I use in the script: xlog(SAVING THE SUBSCRIBER INTO THE LOCATION TABLE,

[OpenSIPS-Users] Presence BLF TLS

Hello I have Grandstream phone with setup BLF button. Opensips use for REGISTER|Presence|BLF When set Grandstream options to Use Actual Ephemeral Port in Contact with TCP/TLS phone add to Contact header port to which it is connected BLF work fine. But if phone set port to standard 5060 Opensips

Re: [OpenSIPS-Users] memory consumed by t_relay

Hi Chen-Che, Looking at the stats: at 10K: dialog:active_dialogs = 21219 dialog:early_dialogs = 1 at 20K: dialog:active_dialogs = 40473 dialog:early_dialogs = 1 As you can see, your active dialogs keep accumulating and consuming memory - this is why you run out of shared mem.

Re: [OpenSIPS-Users] Callpickup scenario help

Hi Miha, So, what you need to do at OpenSIPS level is to be sure that you send the pick up call to the same FS as used for the call you want to pick up. For that I suggest to use some dialog variable (to attach some information to the calls) in combination with get_dialog_info() function (

Re: [OpenSIPS-Users] Registrar not saving received from Path header

Could you try : save(location,$((ff))) ? Do you get any error ? What is your opensips version ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/22/2013 05:49 PM, Nathaniel L Keeling III wrote: Hello Bogdan, I have validated the script and

Re: [OpenSIPS-Users] Slight problem routing 100s and 183s

Hi Nick, I guess you simply have 2 calls in there. The callid : 4737d441-5fb15ea7-7142c0d8@192.168.2.11 comes from .11(phone) goes to .5(opensips) and to .10 (asterisk) - this call is not picked up (there is only a trying from asterisk), so .11 fires a CANCEL which ends the call. I do not

Re: [OpenSIPS-Users] VIA relay error using mhomed=1

No, it is not a retransmission as it is the same process and there is no second set of logs for receiving a message from network: May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:parse_msg: SIP Request: May 20 11:34:52 jkt-svr-mvapp-2 rtsip-service[1411]: DBG:core:parse_msg: method:

Re: [OpenSIPS-Users] Presence BLF TLS

I do not have pcap because use TLS. But I can tomorrow create pcap TCP. Two phone one enable Use Actual Ephemeral Port in Contact with TCP/TLS ( 2...@g-voip.stb.ua ) AOR:: 9...@g-voip.stb.ua Contact:: sip:@10.222.1.253:5060;transport=tls Q=

Re: [OpenSIPS-Users] Presence BLF TLS

The registration info is used only for routing the SUBSCRIBE (which is ok). For the NOTIFY routing, the info from SUBSCRIBE + its 200 OK is used - this is why I need the pcap. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com On 05/22/2013 08:24 PM,

Re: [OpenSIPS-Users] Slight problem routing 100s and 183s

Hello Bogdan, Thank you so much for your response, and your time! The log is for the same call, only, the callid is getting changed by asterisk. What is happening is: 192.168.2.11 (UAC) - 192.168.2.5 (OpenSIPSIn) INVITE Call-ID: 4737d441-5fb15ea7-7142c0d8@192.168.2.11. 192.168.2.5 (OpenSIPSIn)

[OpenSIPS-Users] DO_Routing Jump Timer

Hello, I am using do_rouring module on opensips 1.8.3 I have a problem such as if the trunk is unreachable it will make a lot of delay and the user may send a cancel before it jump to next provider ,how to avoid that and let it jump if there were no response in 4 seconds from the trunk

Re: [OpenSIPS-Users] Do_Routing problem

Dear bogdan, Thanks for you ,my problem been solved Regards khaled From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, May 22, 2013 2:12 PM To: M.Khaled W Chehab Cc: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Do_Routing problem Kaled, You

Re: [OpenSIPS-Users] Do_Routing problem

Mismatched prefix between IDT BICS etc...? N. On 5/22/13, M.Khaled W Chehab kche...@icucall.com wrote: Dear bogdan, Thanks for you ,my problem been solved Regards khaled From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] Sent: Wednesday, May 22, 2013 2:12 PM To: M.Khaled W

[OpenSIPS-Users] Use_next-$avp(gw_id) parameter

Dears I am using opensips 1.8.3 with do_routing module From documentation: The name of the avp for storing the id of the current selected gateway/destination - once a new destination is selected (via the use_next_gw() function), the AVP will be updated with the ID of the new selected

[OpenSIPS-Users] acc record outbound response to originator

I'm wanting to record the response sent back to the call originator, not just the last winning response selected by tm module. There are cases where opensips translates like a received 503 to a 500 or like if you don't trust the response you received from a vendor that is not using sip codes

Re: [OpenSIPS-Users] B2BUA Segfault

Thank you -- I've recompiled and enabled the memory debug. I have the log file from the whole experience available here: http://netbrains-misc.s3.amazonaws.com/opensips/opensips.log (note - real phone numbers and domain names in the log have been replaced with placeholders) The key item seems

Re: [OpenSIPS-Users] Registrar not saving received from Path header

Hello Bogdan, I am using opensips v1.8.3. I was using v1.8.2 earlier but I upgraded thinking it might fix my issue. When I changed the script to the save(location, $((ff))) I get this config error when starting opensips: May 22 17:39:11 [14757] ERROR:core:pv_parse_spec: pvar (inner_name)

Re: [OpenSIPS-Users] Sip server dipping/advice

M.Khaled, It may be easier to perform HTTP Queries to perform the LRN lookup. Have a look at www.bulkvs.com where we offer HTTP based LRN DIPs. Cheers - Daniel -- Forwarded message -- From: M.Khaled W Chehab kche...@icucall.com Date: Tue, May 14, 2013 at 6:17 AM Subject: