Could anyone please help me with changing the ptime value in sipx? We have our VoIP router from bandwidth.com. They changed their ptime setting from 20 to 30 in the mid of december 2010.
Bandwidth also told me the following: 1) Call comes in; Bandwidth requests for ptime of 30 and sipx accepts it. 2) Then sipx sends an invite with a ptime of 20. Please advise, Thanks and have a nice day, SrinivasaRao Seelam ________________________________ From: srinivasa rao <[email protected]> To: Discussion list for users of sipXecs software <[email protected]> Sent: Tue, January 18, 2011 9:38:32 PM Subject: Re: [sipx-users] auto-attendant messages are played real fast You are absolutely correct! I opened a trouble ticket with Bandwidth and they told me that they changed their ptime setting from 20 to 30. Moreover, they made this change in the mid of December 2010. Could you please tell me how I can have the ptime setting changed in the sipx? Bandwidth also told me the following: 1) Call comes in; Bandwidth requests for ptime of 30 and sipx accepts it. 2) Then sipx sends an invite with a ptime of 20. Please advise, Thanks and have a nice day, -SrinivasaRao Seelam ________________________________ From: Tony Graziano <[email protected]> To: Discussion list for users of sipXecs software <[email protected]> Sent: Sun, January 9, 2011 7:04:40 PM Subject: Re: [sipx-users] auto-attendant messages are played real fast Ugh. You are using one of those pesky edgemarc boxes from them. You should be simply doing an unmanaged gateway from sipx to their box. Your should ask them why this is happening. Every one I've ever touched I de-installed and run a firewall in front of it. Here is how to do a call trace. http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer A properly configured firewall (like pfSense) can keep the edgemarc from being needed at all, but I've had many reports of audio issue with bandwidth.com since the start of business on Monday (Jan. 3rd). Is this something that "was working" but started recently also? I suspect they have made some core changes on their network, because we started to see broken 2-way audio, etc., also. On Sun, Jan 9, 2011 at 5:36 PM, srinivasa rao <[email protected]> wrote: We are using Bandwidth.com's equipment. Bandwidth.com gave us a WAN device and VoIP device. > >T1 connected to a WAN device, other end of WAN device is connected to the VoIP >device. Then the VoIP device is connected to the SIPX box. > >I will be glad to run a call trace in sipx. Can you point me to some >documentation that tells me how to run call trace in SIPX? > > >Thanks and have a nice day, > >-SrinivasaRao Seelam > > > > > ________________________________ >From: Tony Graziano <[email protected]> > >To: Discussion list for users of sipXecs software ><[email protected]> >Sent: Sun, January 9, 2011 5:13:38 PM > >Subject: Re: [sipx-users] auto-attendant messages are played real fast > > > > > >On Sun, Jan 9, 2011 at 11:23 AM, srinivasa rao <[email protected]> wrote: > >1) The phones are Polycom IP 430 >> ok 2) Firmware is 3.2.3.1734 > (this firmware is known to have issues, but the calls are not terminating at the phone) 3) When I dial 0 on a IP 430 phone located in our pbx/network to here these auto >attendent messages, they are crystal clear. > good 4) However, when I call the main number from outside, the very same messages are >played real fast. > Do you have more than one gateway/provider to test against? This would really indicate an issue with FS (in sipx), but I have only heard of a couple of issues when the FS system was using "Play_fsv" in freeswitch. Why that is happening is anyone's guess. Perhaps a call trace could shed some light on what is different with the gateway connected call. >Thanks and have a nice day, > >-srinivasarao seelam > > > > >----- Original Message ---- >From: Tony Graziano <[email protected]> > >To: [email protected] >Sent: Sat, January 8, 2011 7:00:37 PM >Subject: Re: [sipx-users] auto-attendant messages are played real fast > >What phone and firmware, this was never answered. >============================ >Tony Graziano, Manager >Telephone: 434.984.8430 >Fax: 434.984.8431 > >Email: [email protected] > >LAN/Telephony/Security and Control Systems Helpdesk: >Telephone: 434.984.8426 >Fax: 434.984.8427 > >Helpdesk Contract Customers: >http://www.myitdepartment.net/gethelp/ > >----- Original Message ----- >From: [email protected] ><[email protected]> >To: Discussion list for users of sipXecs software ><[email protected]> >Sent: Sat Jan 08 18:50:00 2011 >Subject: Re: [sipx-users] auto-attendant messages are played real fast > >I tried that; basically they exactly look the same: I gave the following >outputs > > > >[root@sipx1 prompts]# file autoattendant.wav >autoattendant.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 >bit, >mono 8000 Hz > >[root@sipx1 prompts]# file main.wav >main.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono >8000 Hz >[root@sipx1 prompts]# > > >In the above output, autoattendant.wav came with the installation. And the >main.wav file is the one I uploaded. > >Like I mentioned, it was working great; the problem just started a few weeks >ago. Also, I noticed that any auto attendant message that is upload is >getting >played very fast. > >Kindly advise what should I try next. > >Thanks and have a nice day, > >SrinivasaRao Seelam > > >----- Original Message ---- >From: Douglas Hubler <[email protected]> >To: Discussion list for users of sipXecs software ><[email protected]> >Sent: Wed, January 5, 2011 11:29:20 PM >Subject: Re: [sipx-users] auto-attendant messages are played real fast > >On Wed, Jan 5, 2011 at 4:40 PM, srinivasa rao <[email protected]> wrote: >> We have auto attendants setup in our sipx setup. For the last few weeks, >> these >> auto attendant messages are played very very fast. For example, our main >>message >> >> used to take 40 seconds to complete it, and it takes very less time to >complete >> it. >> >> However, when I play the auto attendant messages manually in the sipx >> administration page, the auto attendent messages are played correctly. >> >> This is a non profit agency, and I would really appreciate your feed back >> in >> resolving the issue. > >Run this command on the *.wav files > file myfile.wav > >Example: > >file tryagain.wav > tryagain.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, >16 bit, mono 8000 Hz > > >it will show you the format details. Check the bitrate and encoding >on a file that come with sipxecs. >_______________________________________________ >sipx-users mailing list >[email protected] >List Archive: http://list.sipfoundry.org/archive/sipx-users/ > >_______________________________________________ >sipx-users mailing list >[email protected] >List Archive: http://list.sipfoundry.org/archive/sipx-users/ >_______________________________________________ >sipx-users mailing list >[email protected] >List Archive: http://list.sipfoundry.org/archive/sipx-users/ > >_______________________________________________ >sipx-users mailing list >[email protected] >List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > > >_______________________________________________ >sipx-users mailing list >[email protected] >List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net Blog: http://blog.myitdepartment.net Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
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