Re: [sipx-users] Polycom Soundpoint cant register

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <42928> Message-ID: Ok I was too quick to post. A google search told me where the bind config files where. /var/named/ I check t

Re: [sipx-users] Polycom Soundpoint cant register

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <5346d54f53154b428cbef30d4879fab2301...@nscmail.nsc.internal> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <42927> Message-ID: mpicher wrote > Well, latest firmware that you wan

Re: [sipx-users] Strange Message

On Thu, 2010-03-04 at 23:05 -0500, Andres Jaramillo wrote: > > Thanks for answer, > > Is this normal ? What's normal? > And why the audioCodes says that One of the basic headers (To, From, > CSeq, Call-Id, Via) is missing in the message ? Well, neither the To nor the From headers have val

Re: [sipx-users] Strange Message

2010/3/4 Scott Lawrence > On Thu, 2010-03-04 at 22:35 -0500, Andres Jaramillo wrote: > > Hi, > > > > > > Someone knows what can mean these messages that is receiving the > > AudioCodes from the SIPx Server ? ,, there are many of these , with > > differents users , I'm seeing this even when nobody

Re: [sipx-users] Strange Message

On Thu, 2010-03-04 at 22:35 -0500, Andres Jaramillo wrote: > Hi, > > > Someone knows what can mean these messages that is receiving the > AudioCodes from the SIPx Server ? ,, there are many of these , with > differents users , I'm seeing this even when nobody is using the > system. ( No calls) >

[sipx-users] Fwd: Strange Message

New message, 19d:3h:42m:3s SUBSCRIBE sip:6062...@192.168.0.122 SIP/2.0 Record-Route: Max-Forwards: 17 Contact: To: "fiduciaria occidente" From: "Deisy Carol¡na Pinto";tag=f8264132 Call-Id: ZjBkNDUzMjQ2MDg0MDEwYTRhNDU3YTdkNWQwY2YzMjM. Cseq: 1 SUBSCRIBE Subject: Expires: 3600 Accept: multipart/rel

Re: [sipx-users] Strange Message

I'm new in all this stuff, sorry (@'.')@ @('.'@) 2010/3/4 Andres Jaramillo > Hi, > > Someone knows what can mean these messages that is receiving the AudioCodes > from the SIPx Server ? ,, there are many of these , with differents users , > I'm seeing this even when nobody is using the system

[sipx-users] Strange Message

Hi, Someone knows what can mean these messages that is receiving the AudioCodes from the SIPx Server ? ,, there are many of these , with differents users , I'm seeing this even when nobody is using the system. ( No calls) 19d:3h:31m:16s SUBSCRIBE sip:6072...@192.168.0.122SIP/2.0 Record-Route: Ma

Re: [sipx-users] IVR hangup after recording name or personal greeting

Peter, I wanted to capture a snapshot. So I set log level to DEBUG and restarted the services. Now I can't reproduce the problem. The IVR lady (Karen?) is no longer rudely hanging up on me after recording name / personal greeting. I swear I wasn't hallucinating when I reported the issue earlier.

Re: [sipx-users] 250 Concurrent Calls?

Much more extensive testing was later performed by our test group and the results were posted somewhere on this list or the development list. You can check out the source code and look in the design document as well. Ranga On Thu, Mar 4, 2010 at 8:11 PM, Matt White wrote: > I think your asking

Re: [sipx-users] 250 Concurrent Calls?

I think your asking for the "250" number from the wiki be qualified. Ranga had does some testing a while back and posted some results in the mailing list. A quick google shows this thread from back in 2008: http://www.mail-archive.com/sipx-...@list.sipfoundry.org/msg00038.html -Matt >>> "Ken

Re: [sipx-users] 250 Concurrent Calls?

I'm mobile responding... Realize your bottlenecks or limitations might be more bandwidth limiting than the architecture. RTP streams are peer to peer between endpoints on internal calls. Remote users and ITSP connections use media relay and sipxbridge, respectively. How you plan it has a great de

Re: [sipx-users] 250 Concurrent Calls?

I appreciate your responses. However, I feel like I'm not explaining myself very well. When you say the system is highly scalable, are there any numbers / specs that show just how scalable it is? Does the 250 number for concurrent calls not include RTP streams? Basically, we need to understand

Re: [sipx-users] 250 Concurrent Calls?

Not necessarily. The system is highly scalable. One bridge makes a single point of failure. HA is a master/distributed but not a master/slave model. Even so, it is more preferable (to me anyway) to make any site or group an "island" when it comes to communications. I find it easier to troubleshoot

Re: [sipx-users] 250 Concurrent Calls?

I might be misunderstanding the capabilities of the system. Can't we use two servers in HA (primary and secondary) with a third as a sipXbridge? I didn't realize each server had to run the proxy services. However, the third server wouldn't have any phones attached - it would be used for PSTN call

Re: [sipx-users] 250 Concurrent Calls?

Understand. I'm not sure how "standalone" it will be. For example, right now ANY server in HA must run a proxy. Voicemail is another that is not standalone. Sipxbridge has nothing to do with internal calls, just itsp connections. Once you get a handle on the requirements and they make sense on how

Re: [sipx-users] Minor UI improvement request

+1 Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 3/4/2010 4:25 PM, Pizza Napoletana wrote: > Using 4.1.6-018058 on centos5 ... > > On the user's web GUI, when you navigate to "My Information" tab, the IMAP > password is displayed in clear text. It would fe

Re: [sipx-users] 250 Concurrent Calls?

I'm trying to understand how the 250 number was generated. We'd want to use a dedicated server for the sipXbridge. Ken From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Thursday, March 04, 2010 4:15 PM To: Ken Fulmer Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx

[sipx-users] Minor UI improvement request

Using 4.1.6-018058 on centos5 ... On the user's web GUI, when you navigate to "My Information" tab, the IMAP password is displayed in clear text. It would feel a bit safer, especially if someone is standing right behind you, to display the password as dots or asterisks. Thanks

Re: [sipx-users] 250 Concurrent Calls?

On Thu, Mar 4, 2010 at 4:52 PM, Ken Fulmer < kenful...@icstechnologysolutions.com> wrote: > The admin guide says the following: > > > > *A single sipXrelay instance can comfortably handle 250 concurrent calls > within acceptable limits of jitter and delay without becoming a bottleneck. > * > > >

Re: [sipx-users] On hold calls drop after one minute

On Thu, Mar 4, 2010 at 4:57 PM, Burden, Mike wrote: > I removed the MOH uri from the phone configurations a couple months ago, > but I must not have tested MOH at the sipXbridge after I did that. I > re-enabled MOH at the sipXbridge, and it seems to be working correctly now. > > > > If the pro

Re: [sipx-users] On hold calls drop after one minute

I removed the MOH uri from the phone configurations a couple months ago, but I must not have tested MOH at the sipXbridge after I did that. I re-enabled MOH at the sipXbridge, and it seems to be working correctly now. If the problems that we had before with MOH don't turn up again in a few da

Re: [sipx-users] Delayed call setup with Flow Route - Wrong field in INVITE

Thank you very much for your replies. I forwarded this info about RFC 3261 and Route header to flowroute. Has anybody successfully used flowroute as an ITSP? -Original Message- From: Scott Lawrence [mailto:scottlawr...@avaya.com] Sent: Thursday, March 04, 2010 1:24 PM To: M. Ranganathan

[sipx-users] 250 Concurrent Calls?

The admin guide says the following: A single sipXrelay instance can comfortably handle 250 concurrent calls within acceptable limits of jitter and delay without becoming a bottleneck. Does anyone know what server specs are required to get this number of calls through the server comfortably?

Re: [sipx-users] On hold calls drop after one minute

On Thu, Mar 4, 2010 at 4:13 PM, Burden, Mike wrote: > In case it helps, the phones are Polycom IP550. > > > Mike Burden > Lynk Systems, Inc > e-mail: m...@lynk.com > Phone: 616-532-4985 > > > > > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun.

Re: [sipx-users] Delayed call setup with Flow Route - Wrong field in INVITE

On Thu, 2010-03-04 at 16:09 -0500, M. Ranganathan wrote: > On Thu, Mar 4, 2010 at 3:57 PM, Marcello Manzardo > wrote: > > I am using 4.0.4-017289 ecs-centos5 > > > > > > > > I have setup flowroute ITSP but experiencing about a 1 min delay until a > > call goes through. > > > > Tech support fro

Re: [sipx-users] On hold calls drop after one minute

In case it helps, the phones are Polycom IP550. Mike Burden Lynk Systems, Inc e-mail: m...@lynk.com Phone: 616-532-4985 -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burden, Mike Sent: Thursday, March 04, 2

Re: [sipx-users] On hold calls drop after one minute

Enabling Music on Hold causes us to immediately be unable to retrieve a call that was placed on hold (the call disconnects as soon as we hit Resume). Changing RTP Keepalive Method from "None" to "Empty Packet" also didn't help (no change in symptoms.) Mike Burden Lynk Systems, Inc e-mail: m...@l

Re: [sipx-users] IVR hangup after recording name or personal greeting

I don't know what the CallPilot UI is. So I assume I am using the standard UI. (I'll mail you a snapshot soon.) I also have a somewhat unrelated voicemail question: If a user is deleted, shouldn't /var/sipxdata/mediaserver/data/mailstore/ be deleted as well? Currently those directories are left b

Re: [sipx-users] Delayed call setup with Flow Route - Wrong field in INVITE

On Thu, Mar 4, 2010 at 3:57 PM, Marcello Manzardo wrote: > I am using  4.0.4-017289   ecs-centos5 > > > > I have setup flowroute ITSP but experiencing about a 1 min delay until a > call goes through. > > Tech support from flowroute pointed to a wrong field in the SIP INVITE that > is generated by

[sipx-users] Delayed call setup with Flow Route - Wrong field in INVITE

I am using 4.0.4-017289 ecs-centos5 I have setup flowroute ITSP but experiencing about a 1 min delay until a call goes through. Tech support from flowroute pointed to a wrong field in the SIP INVITE that is generated by sipX. The field in question is the "Route" field that is not supposed

Re: [sipx-users] Outgoing per-user Caller-ID is not respected

On Thu, 2010-03-04 at 21:27 +0100, Pavel Arnošt wrote: > > From header is from "bridge = ITSP" dialog. I (who am not quite expert) think that configuration looks ok. To debug this, you'll need to trace the message flow and see what's going wrong. See: http://wiki.sipfoundry.org/display/xecsus

Re: [sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

On Thu, 2010-03-04 at 11:52 -0500, Michael Scheidell wrote: > > > On 3/4/10 11:40 AM, Michael Scheidell wrote: > > > Create a custom rule, give it local permissions. Prefix is "411" > > > and "0" digits, resulting call is > > > > > > 800-466-4411 and append nothing, choose the gateway and activ

[sipx-users] IVR hangup after recording name or personal greeting

Using 4.1.6-018058 on centos5 ... After a user records and accepts their name or personal greeting by dialing the voicemail extension, the call gets disconnected immediately. Is this the expected behavior? I was hoping that the IVR lady would say something nice like "Your name / personal gree

Re: [sipx-users] Outgoing per-user Caller-ID is not respected

Hi, From header is from "bridge = ITSP" dialog. Sipxbridge.xml looks like: http://www.sipfoundry.org/sipX/schema/xml/sipxbridge-00-00";> EXTERNAL_IP_ADDRESS EXTERNAL_IP_ADDRESS 5080 EXTERNAL_IP_ADDRESS 5090 DOMAIN pbx.DOMAIN 8092 stun01.sipphone.com 20

Re: [sipx-users] On hold calls drop after one minute

On Thu, 2010-03-04 at 10:50 -0500, Burden, Mike wrote: > We are having a problem where calls that are one hold are dropped > after one minute. > > I've put the sipxtrace on our FTP server at: > ftp://ftp.lynk.com/losthold.xml The call is terminated by the BYE at frame 316, which is indeed 1 minut

Re: [sipx-users] Outgoing per-user Caller-ID is not respected

On Thu, 2010-03-04 at 19:55 +0100, Pavel Arnošt wrote: > > in our setup, outgoing per-user Caller-ID is not respected. Outgoing > calls goes through ITSP SIP trunk. Default Caller ID of SIP trunk is > set to "X" number, Caller ID of user is set to "X0003" > number. In this setup, all

Re: [sipx-users] Fw: Default attendant and ACD not working

On Thu, 2010-03-04 at 19:22 +0100, Angelo Turetta wrote: > On 02/03/2010 15:43, Tony Graziano wrote: > > sipxproc --state > > (to see what is running or enabled/disabled) > > doesn't look well > > #sipxproc --state > /usr/lib/ruby/1.8/openssl/ssl.rb:91:in `post_connection_check': hostname >

[sipx-users] Outgoing per-user Caller-ID is not respected

Hi, in our setup, outgoing per-user Caller-ID is not respected. Outgoing calls goes through ITSP SIP trunk. Default Caller ID of SIP trunk is set to "X" number, Caller ID of user is set to "X0003" number. In this setup, all outgoing calls have "X" number in From header, I w

Re: [sipx-users] Fw: Default attendant and ACD not working

On 02/03/2010 15:43, Tony Graziano wrote: > sipxproc --state > (to see what is running or enabled/disabled) doesn't look well #sipxproc --state /usr/lib/ruby/1.8/openssl/ssl.rb:91:in `post_connection_check': hostname not match with the server certificate (OpenSSL::SSL::SSLError) fro

[sipx-users] Concurrent Call Volume

The admin guide says the following: "*Has good media performance: sipXbridge anchors media and is implemented as an efficient media relay service. A single sipXrelay instance can comfortably handle 250 concurrent calls within acceptable limits of jitter and delay without becoming a bottleneck."

[sipx-users] Fwd: CallController

Forgot to cc list. -- Forwarded message -- From: M. Ranganathan Date: Thu, Mar 4, 2010 at 12:30 PM Subject: Re: [sipx-users] CallController To: Kyle Haefner On Thu, Mar 4, 2010 at 11:18 AM, Kyle Haefner wrote: > Hi All, > I can't seem to figure out how to use the call control

Re: [sipx-users] Polycom Soundpoint cant register

I would double check your DNS configuration (and DHCP parameters) as stated. If the appropriate SRV or A records are not created properly, phones will not register. -Under System > Servers > yoursipxecsserver.com are all services running properly? In your AC MP114 does it show your gateway regis

Re: [sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

On 3/4/10 11:40 AM, Michael Scheidell wrote: Create a custom rule, give it local permissions. Prefix is "411" and "0" digits, resulting call is 800-466-4411 and append nothing, choose the gateway and activate it after moving it to the appropriate place in your dialplan. taking tony's sugge

Re: [sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

On 3/4/10 11:04 AM, Tony Graziano wrote: This is one aspect of: http://track.sipfoundry.org/browse/XX-6358 and should clearly be fixed. In the meantime I use this rule: Create a custom rule, give it local permissions. Prefix is "411" and "0" digits, resulting call is 800-466-4411

[sipx-users] CallController

Hi All, I can't seem to figure out how to use the call controller through rest. From the plugin.xml: "Invoke this using the URI /callcontroller/{callingParty}/{calledParty}?agentId" I am assuming that callingParty is a defined user. calledParty is the number to dial What is the agentId? Anyone

Re: [sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

Easy there, big fella! In case you did not notice, I was trying to help. If what I wrote does not help you then politely discard it or ask for clarification. > > you didn't read my email, did you? > > do you know WHY I don't want to do that? > > do you know WHY its important ? > > did you kn

Re: [sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

On Thu, Mar 4, 2010 at 11:02 AM, Scott Lawrence wrote: > On Thu, 2010-03-04 at 10:06 -0500, Michael Scheidell wrote: > > I was trying to take advantage of google 411 (redirecting anyone who > > calls 411) and thought the easiest place to put this is in dial plan. > > > > I created a custom dial pl

Re: [sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

On Thu, 2010-03-04 at 10:06 -0500, Michael Scheidell wrote: > I was trying to take advantage of google 411 (redirecting anyone who > calls 411) and thought the easiest place to put this is in dial plan. > > I created a custom dial plan, 411, and tried to put this in resulting > call[] > 18004664..

[sipx-users] On hold calls drop after one minute

Good morning, We are having a problem where calls that are one hold are dropped after one minute. I've put the sipxtrace on our FTP server at: ftp://ftp.lynk.com/losthold.xml I'm not expert enough at reading them to pick out the root cause of the problem. Can someone take a look? Mike Bur

Re: [sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

you didn't read my email, did you? do you know WHY I don't want to do that? do you know WHY its important ? did you know that SIP URL's are allowed to include domains? On 3/4/10 10:16 AM, Robert Joly wrote: Do not put the domain in your resulting call, only digits belong there. Put 1800466

Re: [sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

Do not put the domain in your resulting call, only digits belong there. Put 18004664411 in the resulting call - this digit string will be sent to whatever gateway you have configured for this dialplan. bob From: sip

[sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

I was trying to take advantage of google 411 (redirecting anyone who calls 411) and thought the easiest place to put this is in dial plan. I created a custom dial plan, 411, and tried to put this in resulting call[] 18004664...@tf.voipmich.com I got this error: Only a limited set of characters

Re: [sipx-users] [sipX-dev] sipxbridge and remote users without server behind nat

Interesting to hear that sipXecs works better for you when behind a NAT. I would like to take a look at a network trace of your system to understand why you are not getting the audio on VM calls. Can you run 'tcpdump -n -nn -s 0 -i any -w vm_no_audio.pcap' on your sipXecs, reproduce the problem an

Re: [sipx-users] which two port fxs for sipx?

Are you asking about a different ATA (analog terminal adapter) device to connect the analog phones to sipx? If so, I have had good success with the Linksys SPA2102. The have 2 FXS ports, and 2 100MB ethernet ports. If you use them, please note that the Linksys device config files still have

Re: [sipx-users] Polycom Soundpoint cant register

On Thu, Mar 4, 2010 at 5:53 AM, Picher, Michael wrote: > Well, latest firmware that you want to use with 4.0.4 is 3.1.3c (split) > with 4.2.0 bootrom. > > I'm not sure that this firmware will work on the 500 series phones > however... this may be part of the problem. > > Aside from that, you have

Re: [sipx-users] Polycom Soundpoint cant register

Well, latest firmware that you want to use with 4.0.4 is 3.1.3c (split) with 4.2.0 bootrom. I'm not sure that this firmware will work on the 500 series phones however... this may be part of the problem. Aside from that, you haven't give us much to go off. As far as noob problems go, make sure y

[sipx-users] Polycom Soundpoint cant register

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <42831> Message-ID: Hi This is my first try at setting up a SipXecs pbx. Its for a small office, only 8 phones(all polycom soundpoint 500) and 1