Getting 407 Proxy Authentication Required when dialing site to site

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Steven Lane

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Sep 29, 2015, 5:33:06 PM9/29/15
to sipxcom-users
Hi Everyone,
I wanted to see if I could get some help here because for the last 2 days I have been racking my head against the wall about site to site sipxcom calling. (Also known as sipxcom to sipxcom calling.) Let me first explain my set up:

3 server cluster of sipX 4.4 in production
5 server cluster of sipXCom 15.08 in staging (testing) between 2 geographically dispersed data centers

The 3 server cluster in production is only at one data center. I know it is an older version, but shouldn't I ba able to call site to site with the same set up?
Anyhow, I set up the sipXCom 15.08 5 server cluster to be able to call the 3 server cluster but I am getting a 408 in the CDRs and when I ngrep on the server I see a 407 when dialing from my phone to the 15.08 cluster. Below is an ngrep of the sip dialog:

INVITE sip:55...@sips1.domain.com:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
From: "Steven (Test)" <sip:10...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:55...@sips1.domain.com;user=phone>.
CSeq: 1 INVITE.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.6.0314.
Accept-Language: en.
Supported: 100rel,replaces.
Allow-Events: talk,hold,conference.
Max-Forwards: 70.
Content-Type: application/sdp.
Content-Length: 294.
.
v=0.
o=- 1443559791 1443559791 IN IP4 10.10.28.119.
s=Polycom IP Phone.
c=IN IP4 10.10.28.119.
t=0 0.
a=sendrecv.
m=audio 2256 RTP/AVP 9 0 8 18 101.
a=rtpmap:9 G722/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.

#
SIP/2.0 100 Trying.
From: "Steven (Test)" <sip:10...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:55...@sips1.domain.com;user=phone>.
Cseq: 1 INVITE.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
Content-Length: 0.
.

#
SIP/2.0 407 Proxy Authentication Required.
From: "Steven (Test)" <sip:10...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:55...@sips1.domain.com;user=phone>;tag=AZiXtP.
Cseq: 1 INVITE.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
Proxy-Authenticate: Digest realm="sips1.domain.com", nonce="ca84cce5e343e472a813e50675726313560af972", qop="auth".
Server: sipXecs/15.08 sipXecs/sipXproxy (Linux).
Date: Tue, 29 Sep 2015 20:49:54 GMT.
Content-Length: 0.
.

#
ACK sip:55...@sips1.domain.com:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
From: "Steven (Test)" <sip:10...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:55...@sips1.domain.com;user=phone>;tag=AZiXtP.
CSeq: 1 ACK.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.6.0314.
Accept-Language: en.
Max-Forwards: 70.
Content-Length: 0.

The call obviously goes fast busy. To add to the mix here, I am using Polycom 550 phones. My goal is to get the two systems to be able to call and transfer calls to each other if that is possible. I really appreciate any help here. 

Thanks in advance,

Steven L.

Todd Hodgen

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Sep 30, 2015, 12:36:42 AM9/30/15
to Steven Lane, sipxcom-users

Have you added the subnet for the far end into your “Internet Settings” on the local configuration.

 

Todd R. Hodgen

President / Founder

Sound IP Telecom

http://SIPTelecom.systems

206-432-4344 - Direct

206-390-4689 - Cell

 

SIP Telecom Logo1 copy

Your Puget Sound Telcom Partner

 

Do you Like Snow, Rain, Sleet, Hail, Thunder, Lightning?

If you do, then you will love the Cloud!

 

We appreciate your business referrals!

A Division of Misiu Systems LLC

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Steven Lane

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Sep 30, 2015, 12:37:13 PM9/30/15
to Todd Hodgen, sipxcom-users
Thanks for the reply Todd. I think you are asking if I have added the subnet in the " Internet Calling" tab. As you can see I have the following:

Inline image 1

That is a picture of the far end "Internet Calling" settings which is a sipX 4.4 cluster. That is what is in production and what I am trying to call. The IPs of both clusters are:
sipXCom 15.08 
Inline image 2

sipXecs 4.4 (production cluster)
Inline image 3

So I should be able to call between the two clusters. Right? When I trained with Mike P. at eZuce we did it so facst and easy, and I even followed the document that was written for the training explaining how to do it. Still no joy.

Steven Lane
Senior VoIP Engineer Telephony Systems Engineering
PennyMac
114800 Trinity Blvd. 
Fort Worth, TX 76155
O (469) 629-6157 | Ext 5394

Steven Lane

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Oct 1, 2015, 4:18:08 PM10/1/15
to sipxcom-users
Can I call this broken since I can't even get sipXCom to sipXCom working between the same versions of sipXCom?

Steven Lane

On Tuesday, September 29, 2015 at 4:33:06 PM UTC-5, Steven Lane wrote:
Hi Everyone,
I wanted to see if I could get some help here because for the last 2 days I have been racking my head against the wall about site to site sipxcom calling. (Also known as sipxcom to sipxcom calling.) Let me first explain my set up:

3 server cluster of sipX 4.4 in production
5 server cluster of sipXCom 15.08 in staging (testing) between 2 geographically dispersed data centers

The 3 server cluster in production is only at one data center. I know it is an older version, but shouldn't I ba able to call site to site with the same set up?
Anyhow, I set up the sipXCom 15.08 5 server cluster to be able to call the 3 server cluster but I am getting a 408 in the CDRs and when I ngrep on the server I see a 407 when dialing from my phone to the 15.08 cluster. Below is an ngrep of the sip dialog:

INVITE sip:55...@sips1.domain.com:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
From: "Steven (Test)" <sip:10...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:55...@sips1.domain.com;user=phone>.
CSeq: 1 INVITE.
Cseq: 1 INVITE.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
Content-Length: 0.
.

#
SIP/2.0 407 Proxy Authentication Required.
From: "Steven (Test)" <sip:10...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:55...@sips1.domain.com;user=phone>;tag=AZiXtP.
Cseq: 1 INVITE.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
Proxy-Authenticate: Digest realm="sips1.domain.com", nonce="ca84cce5e343e472a813e50675726313560af972", qop="auth".
Server: sipXecs/15.08 sipXecs/sipXproxy (Linux).
Date: Tue, 29 Sep 2015 20:49:54 GMT.
Content-Length: 0.
.

#
ACK sip:55...@sips1.domain.com:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
From: "Steven (Test)" <sip:10...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:55...@sips1.domain.com;user=phone>;tag=AZiXtP.
CSeq: 1 ACK.

Matthew Kitchin

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Oct 1, 2015, 4:36:39 PM10/1/15
to sipxco...@googlegroups.com
Can you post a pcap? Any SBCs or other equipment involved? Sipxbridge?
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Steven Lane

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Oct 1, 2015, 5:24:14 PM10/1/15
to sipxcom-users
There are no other pieces involved. Just dialing an extension from one sipXCom cluster to the other. I can try to do a pcap, but I did not install a homer server in the cluster. I did an ngrep from the first server and posted it here, but any other suggestions as to getting the pcap would be highly appreciated. 

--Steven L.


On Thursday, October 1, 2015 at 1:36:39 PM UTC-7, Matthew Kitchin wrote:
Can you post a pcap? Any SBCs or other equipment involved? Sipxbridge?

On 10/1/2015 3:18 PM, Steven Lane wrote:
Can I call this broken since I can't even get sipXCom to sipXCom working between the same versions of sipXCom?

Steven Lane

On Tuesday, September 29, 2015 at 4:33:06 PM UTC-5, Steven Lane wrote:
Hi Everyone,
I wanted to see if I could get some help here because for the last 2 days I have been racking my head against the wall about site to site sipxcom calling. (Also known as sipxcom to sipxcom calling.) Let me first explain my set up:

3 server cluster of sipX 4.4 in production
5 server cluster of sipXCom 15.08 in staging (testing) between 2 geographically dispersed data centers

The 3 server cluster in production is only at one data center. I know it is an older version, but shouldn't I ba able to call site to site with the same set up?
Anyhow, I set up the sipXCom 15.08 5 server cluster to be able to call the 3 server cluster but I am getting a 408 in the CDRs and when I ngrep on the server I see a 407 when dialing from my phone to the 15.08 cluster. Below is an ngrep of the sip dialog:

INVITE sip:55...@sips1.domain.com:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
From: "Steven (Test)" <sip:...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:...@sips1.domain.com;user=phone>.
CSeq: 1 INVITE.
From: "Steven (Test)" <sip:...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:...@sips1.domain.com;user=phone>.
Cseq: 1 INVITE.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
Content-Length: 0.
.

#
SIP/2.0 407 Proxy Authentication Required.
From: "Steven (Test)" <sip:...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:...@sips1.domain.com;user=phone>;tag=AZiXtP.
Cseq: 1 INVITE.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
Proxy-Authenticate: Digest realm="sips1.domain.com", nonce="ca84cce5e343e472a813e50675726313560af972", qop="auth".
Server: sipXecs/15.08 sipXecs/sipXproxy (Linux).
Date: Tue, 29 Sep 2015 20:49:54 GMT.
Content-Length: 0.
.

#
ACK sip:55...@sips1.domain.com:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
From: "Steven (Test)" <sip:...@sips1.domain.com>;tag=3CDA156F-E168FDF4.
To: <sip:...@sips1.domain.com;user=phone>;tag=AZiXtP.
CSeq: 1 ACK.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.6.0314.
Accept-Language: en.
Max-Forwards: 70.
Content-Length: 0.

The call obviously goes fast busy. To add to the mix here, I am using Polycom 550 phones. My goal is to get the two systems to be able to call and transfer calls to each other if that is possible. I really appreciate any help here. 

Thanks in advance,

Steven L.

Joe Micciche

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Oct 1, 2015, 6:33:45 PM10/1/15
to sipxco...@googlegroups.com
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

On 10/01/2015 05:24 PM, Steven Lane wrote:
> There are no other pieces involved. Just dialing an extension from
> one sipXCom cluster to the other. I can try to do a pcap, but I did
> not install a homer server in the cluster. I did an ngrep from the
> first server and posted it here, but any other suggestions as to
> getting the pcap would be highly appreciated.

On each server:

You need to ID your network interface with ifconfig; then tcpdump on
each server to capture packets to/from the /other/ server:
- --
tcpdump -i eth0 -s 0 -w server1.pcap host 1.2.3.4
tcpdump -i eth0 -s 0 -w server2.pcap host 5.6.7.8

I set up sipX-to-sipX calling via a Custom Dialing rule, assign it a
minimum permission (e.g. "Internal") which all phones have, and point
each cluster to the other. Works fine.

joe
>>> "Steven (Test)" <sip:...@sips1.domain.com <javascript:>
>>>> ;tag=3CDA156F-E168FDF4.
>>> To: <sip:...@sips1.domain.com <javascript:>;user=phone>. CSeq:
>>> 1 INVITE. Call-ID: 9e516d4b-8498...@10.10.28.119
>>> <javascript:>. Contact: <sip:...@10.10.28.119 <javascript:>>.
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE,
>>> SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent:
>>> PolycomSoundPointIP-SPIP_550-UA/3.2.6.0314. Accept-Language:
>>> en. Supported: 100rel,replaces. Allow-Events:
>>> talk,hold,conference. Max-Forwards: 70. Content-Type:
>>> application/sdp. Content-Length: 294. . v=0. o=- 1443559791
>>> 1443559791 IN IP4 10.10.28.119. s=Polycom IP Phone. c=IN IP4
>>> 10.10.28.119. t=0 0. a=sendrecv. m=audio 2256 RTP/AVP 9 0 8 18
>>> 101. a=rtpmap:9 G722/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8
>>> PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no.
>>> a=rtpmap:101 telephone-event/8000.
>>>
>>> # U 10.4.22.51:5060 -> 10.10.28.119:5060 SIP/2.0 100 Trying.
>>> From: "Steven (Test)" <sip:...@sips1.domain.com <javascript:>
>>>> ;tag=3CDA156F-E168FDF4.
>>> To: <sip:...@sips1.domain.com <javascript:>;user=phone>.
>>> Call-Id: 9e516d4b-8498...@10.10.28.119 <javascript:>. Cseq: 1
>>> INVITE. Via: SIP/2.0/UDP
>>> 10.10.28.119;branch=z9hG4bK83f24bc59FD06032. Content-Length:
>>> 0. .
>>>
>>> # U 10.4.22.51:5060 -> 10.10.28.119:5060 SIP/2.0 407 Proxy
>>> Authentication Required. From: "Steven (Test)"
>>> <sip:...@sips1.domain.com <javascript:>
>>>> ;tag=3CDA156F-E168FDF4.
>>> To: <sip:...@sips1.domain.com
>>> <javascript:>;user=phone>;tag=AZiXtP. Call-Id:
>>> 9e516d4b-8498...@10.10.28.119 <javascript:>. Cseq: 1 INVITE.
>>> Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.
>>> Proxy-Authenticate: Digest realm="sips1.domain.com",
>>> nonce="ca84cce5e343e472a813e50675726313560af972", qop="auth".
>>> Server: sipXecs/15.08 sipXecs/sipXproxy (Linux). Date: Tue, 29
>>> Sep 2015 20:49:54 GMT. Content-Length: 0. .
>>>
>>> # U 10.10.28.119:5060 -> 10.4.22.51:5060 ACK
>>> sip:55...@sips1.domain.com:5060;user=phone SIP/2.0. Via:
>>> SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032. From:
>>> "Steven (Test)" <sip:...@sips1.domain.com <javascript:>
>>>> ;tag=3CDA156F-E168FDF4.
>>> To: <sip:...@sips1.domain.com
>>> <javascript:>;user=phone>;tag=AZiXtP. CSeq: 1 ACK. Call-ID:
>>> 9e516d4b-8498...@10.10.28.119 <javascript:>. Contact:
>>> <sip:...@10.10.28.119 <javascript:>>. Allow: INVITE, ACK, BYE,
>>> CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK,
>>> UPDATE, REFER. User-Agent:
>>> PolycomSoundPointIP-SPIP_550-UA/3.2.6.0314. Accept-Language:
>>> en. Max-Forwards: 70. Content-Length: 0.
>>>
>>> The call obviously goes fast busy. To add to the mix here, I am
>>> using Polycom 550 phones. My goal is to get the two systems to
>>> be able to call and transfer calls to each other if that is
>>> possible. I really appreciate any help here.
>>>
>>> Thanks in advance,
>>>
>>> Steven L.
>>>
>> -- You received this message because you are subscribed to the
>> Google Groups "sipxcom-users" group. To unsubscribe from this
>> group and stop receiving emails from it, send an email to
>> sipxcom-user...@googlegroups.com <javascript:>. To post to this
>> group, send email to sipxco...@googlegroups.com <javascript:>.
>> Visit this group at
>> http://groups.google.com/group/sipxcom-users. To view this
>> discussion on the web visit
>> <https://groups.google.com/d/msgid/sipxcom-users/14772149-1b50-46f4-8a44-6aaf3045e0aa%40googlegroups.com?utm_medium=email&utm_source=footer>
>>
>>
https://groups.google.com/d/msgid/sipxcom-users/14772149-1b50-46f4-8a44-6aaf3045e0aa%40googlegroups.com
>> . For more options, visit https://groups.google.com/d/optout.
>>
>>
>>
>

- --
==================================================================
Joe Micciche | Red Hat, Inc. | www.redhat.com
Senior Communications Engineer | +1.919.754.4554 | X81 44554
==================================================================
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Todd Hodgen

unread,
Oct 1, 2015, 11:37:38 PM10/1/15
to Steven Lane, sipxcom-users

I have recently turned the feature up between two sites – running sipxcom 15.04.   Hi s has been working without issue.  You may have run across a bug of some type, or possibly a router issues, or several other things.   Personally, I don’t believe it is broken.

 

Have you tried to do tcpdump on the interface of the different sites when you try calls to see if it is being routed from end to end correctly, and then look at what is arriving at each end?  That would probably be my next step.

 

 

I have several other sites using it today as well, on various versions of sipxcom.

 

Todd R. Hodgen

President / Founder

Sound IP Telecom

http://SIPTelecom.systems

206-432-4344 - Direct

206-390-4689 - Cell

 

SIP Telecom Logo1 copy

Your Puget Sound Telcom Partner

 

Do you Like Snow, Rain, Sleet, Hail, Thunder, Lightning?

If you do, then you will love the Cloud!

 

We appreciate your business referrals!

A Division of Misiu Systems LLC

 

From: sipxco...@googlegroups.com [mailto:sipxco...@googlegroups.com] On Behalf Of Steven Lane


Sent: Thursday, October 01, 2015 1:18 PM
To: sipxcom-users <sipxco...@googlegroups.com>

Subject: [sipxcom-users] Re: Getting 407 Proxy Authentication Required when dialing site to site

Can I call this broken since I can't even get sipXCom to sipXCom working between the same versions of sipXCom?

 

Steven Lane


On Tuesday, September 29, 2015 at 4:33:06 PM UTC-5, Steven Lane wrote:

Hi Everyone,

I wanted to see if I could get some help here because for the last 2 days I have been racking my head against the wall about site to site sipxcom calling. (Also known as sipxcom to sipxcom calling.) Let me first explain my set up:

 

3 server cluster of sipX 4.4 in production

5 server cluster of sipXCom 15.08 in staging (testing) between 2 geographically dispersed data centers

 

The 3 server cluster in production is only at one data center. I know it is an older version, but shouldn't I ba able to call site to site with the same set up?

Anyhow, I set up the sipXCom 15.08 5 server cluster to be able to call the 3 server cluster but I am getting a 408 in the CDRs and when I ngrep on the server I see a 407 when dialing from my phone to the 15.08 cluster. Below is an ngrep of the sip dialog:

 

Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.

From: "Steven (Test)" <sip:10...@sips1.domain.com>;tag=3CDA156F-E168FDF4.

To: <sip:55...@sips1.domain.com;user=phone>.

CSeq: 1 INVITE.

Cseq: 1 INVITE.

Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.

Content-Length: 0.

.

 

#

SIP/2.0 407 Proxy Authentication Required.

From: "Steven (Test)" <sip:10...@sips1.domain.com>;tag=3CDA156F-E168FDF4.

To: <sip:55...@sips1.domain.com;user=phone>;tag=AZiXtP.

Cseq: 1 INVITE.

Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.

Proxy-Authenticate: Digest realm="sips1.domain.com", nonce="ca84cce5e343e472a813e50675726313560af972", qop="auth".

Server: sipXecs/15.08 sipXecs/sipXproxy (Linux).

Date: Tue, 29 Sep 2015 20:49:54 GMT.

Content-Length: 0.

.

 

#

Via: SIP/2.0/UDP 10.10.28.119;branch=z9hG4bK83f24bc59FD06032.

From: "Steven (Test)" <sip:10...@sips1.domain.com>;tag=3CDA156F-E168FDF4.

To: <sip:55...@sips1.domain.com;user=phone>;tag=AZiXtP.

CSeq: 1 ACK.

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER.

User-Agent: PolycomSoundPointIP-SPIP_550-UA/3.2.6.0314.

Accept-Language: en.

Max-Forwards: 70.

Content-Length: 0.

 

The call obviously goes fast busy. To add to the mix here, I am using Polycom 550 phones. My goal is to get the two systems to be able to call and transfer calls to each other if that is possible. I really appreciate any help here. 

 

Thanks in advance,

 

Steven L.

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