NOW that google voice has pulled the plug from Simonics, and some OBI ATA'S, a lot of us would like to flash our ATA to accommodate and connect directly to the SO CALLED new sip standard google voice is now using. More or less convert the ATA SPA 2102 to an obi 202 . Anybody have the know how ??? And give us some options instead of being force to buy the only other box from polycom !!!!!! http://www.polycom.com/voice-conferencing-solutions/voip-adapters/obi202.html
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[Equipment] spa2102 ATA hack for google voice
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Obihai 110 - PhonePower
First post:
Been running an OBI110 using PhonePower since the first upgrade extortion. Pretty satisfied with their service and the OBI. But decided to upgrade to the 202. Considering the recent announcements, figured it was time earlier this week. Followed the directions on Obitalk. First was to delete the 110 device. Should never have done that, but whatever. Installed the 202, and using Obitalk went to configure the PhonePower account, and did not find the old automatic configuration, like current Anveo. But, went into other providers and followed the instructions. Device would not register. Now, no phone.
Contacted PhonePower. 45 minutes on hold. Took my name. Emailed me an updated set of configuration settings. Tried them. They failed. Emailed PhonePower (PP). They sent a different set. They failed.
Called them the next day. Initial customer rep acted very frustrated with me for telling her that I had repeatedly tried everything they sent me. We went through it again and again. The settings they gave me would not save in the device. So, we had to go through them a few times. She got frustrated enough to send me to a real tech, who began a four hour session to first try to get the 202 to work, and then to get my 110 back to work.
He was able, with various settings, mostly the port setting, to get the 202 to register. But there was still a problem with the firmware. On PP's account dashboard, you could see that the OBI was generating extra user registrations within minutes of the OBI registering the service. PP has a limit of 4 within any four hour period. So we could get the 202 working. But it would only work for 5-10 minutes after registering.
We went through both the Obitalk configuration first, then the direct IP address link for the OBI settings. That did not seem to make a difference.
This is where I'm not clear on exactly the problem. The PP tech said the problem was with the current firmware. We we tried rolling it back to known PP-OBI functioning firmware, every upload came back with a "checksum error" and would not install. We could not test them. Hence we rolled my install back to the 110, and after an hour got it working again.
I contacted OBI via their Facebook messenger customer service account. 24+ hours later a text popped up on Messenger. I tried to explain the situation. Their rep had no idea what I was talking about. I referred them to PP, and the agent, and this board, to try to find a solution. The OBI rep essentially blew me off. Really changed my opinion of them. But, I still want to get the 202 box functioning. I'm paid up for a year with PhonePower, like the quality of their connection, and want to stay with them. But I have no idea what to do next. I have a year of support with my OBI. First install was last week, and they simply don't give a rats about honoring it. They basically said, "We hope you get your Phonepower issue resolved". No referral to any real assistance. They won't respond to any of my texts on the thread.
Anyone got a suggestion on how to proceed? I'd like to get it working with the right firmware and settings.
Thanks,
mws
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GVSIP configuration helper for FreePBX users
This is a very simple script to help if you are creating multiple GVSIP accounts in FreePBX. You must replace the three items at the top of the file with values you use for oAuth Client ID and Client Secret (you only need one working pair of those, no matter how many accounts you have), and also a one word name for your system (can be anything, when you run it once you'll see how it's used). It then walks you through getting an oAuth Refresh Token (instructions mostly adapted from RonR's previous post), asks for a bit of additional information, and then gives you a suggested configuration. There is no error or sanity checking; "Garbage In - Garbage Out" applies here. This sort of assumes you have created at least one account already and therefore the only configuration information you need is for the new accounts. If you have never created any GVSIP account before, you should probably follow naf's instructions, at least for the [global] and [transport_tls] contexts (and yet I know that theoretically you can set some of this from the FreePBX GUI, but if you have PJSIP extensions doing that may cause problems, at least with regard to the tls stuff). And also you may need to follow RonR's instructions to get an oAuth Client ID and Client Secret, if you don't already have those.
I do things a little differently than others who have posted scripts and I think you will find that if you have several accounts, the way I do things makes it a little bit easier to keep track of your accounts. It works for me, but you are of course free to modify it as you see fit. No error checking, bells, whistles, or frills. Give it a dry run with dummy data to get a sense of what it does. If you're not satisfied, feel free to return it for a full refund. :D
One hint to others who write scripts: No, you really don't need any added dialplan in extensions_custom.conf, not to make GVSIP work anyway. If you are using lines of custom dialplan to strip the +1 from the start of the incoming Google Voice number then forget about it and run this script once and it will show you the right way to do that, using the from-pstn-e164-us context in your trunk settings.
EDIT: Added reminders to click Submit after creating pages in the FreePBX GUI.
#!/bin/bashoAuthClientID="PUT YOUR OAUTH CLIENT ID HERE"oAuthClientSecret="PUT YOUR OAUTH CLIENT SECRET HERE"SystemName="PUT A ONE WORD NAME FOR YOUR SYSTEM HERE (NO SPACES!!!)"echo ""echo ""echo "GVSIP Config Helper"echo ""echo ""echo "Instructions to get Refresh Token:"echo ""echo ""echo "1. Log into the Google Voice account for which you need a token"echo ""echo "2. Go to https://developers.google.com/oauthplayground"echo ""echo "3. Click the gear icon in the top right of the page"echo ""echo "4. Check Use your own oAuth credentials"echo ""echo "5. Enter oAuth Client ID: $oAuthClientID"echo " and oAuth Client secret: $oAuthClientSecret"echo ""echo "6. Click Close"echo ""echo "7. Enter https://www.googleapis.com/auth/googletalk at Input your own scopes"echo ""echo "8. Click Authorize API"echo ""echo "9. Sign into your Google Voice account again if you are prompted to do so"echo ""echo "10. Click Allow"echo ""echo "11. Click Exchange authorization code for tokens"echo ""echo -n "12. Record the Refresh Token (without the quotation marks) in a safe place and enter it here: "read oAuthRefreshTokenecho ""echo ""echo "Now please enter the following additional information:"echo ""echo -n "Enter the Google Voice account name (the part before @gmail.com): "read AccountNameecho ""echo -n "Enter the 10 digit Google Voice telephone number: "read GVPhoneNumberecho ""echo -n "Enter the account user's name with NO SPACES: "read UserFirstNameecho ""echo -n "Enter the city associated with the Google Voice telephone number, with NO SPACES: "read PrefixCityecho ""echo ""echo "Add the following to the file /etc/asterisk/pjsip_custom_post.conf:"echo ""echo "[gvsip-$AccountName]"echo "type=registration"echo "outbound_auth=gvsip-$AccountName"echo "server_uri=sip:obihai.sip.google.com"echo "outbound_proxy=sip:obihai.telephony.goog:5061\;transport=tls\;lr\;hide"echo "client_uri=sip:$AccountName$GVPhoneNumber@obihai.sip.google.com"echo "retry_interval=60"echo "support_path=yes"echo "support_outbound=yes"echo "line=yes"echo "endpoint=gvsip-$AccountName"echo "contact_additional_params=obn=$SystemName-$UserFirstName-$PrefixCity"echo "transport=transport_tls"echo "transport_reuse=no"echo "contact_user=$GVPhoneNumber"echo ""echo "[gvsip-$AccountName]"echo "type=auth"echo "auth_type=oauth"echo "refresh_token=$oAuthRefreshToken"echo "oauth_clientid=$oAuthClientID"echo "oauth_secret=$oAuthClientSecret"echo "username=$AccountName$GVPhoneNumber"echo "realm=obihai.sip.google.com"echo ""echo "[gvsip-$AccountName]"echo "type=aor"echo "contact=sip:obihai.sip.google.com"echo ""echo "[gvsip-$AccountName]"echo "type=endpoint"echo "context=from-pstn-e164-us"echo "disallow=all"echo "allow=ulaw"echo "allow=opus"echo "outbound_auth=gvsip-$AccountName"echo "outbound_proxy=sip:obihai.telephony.goog:5061\;transport=tls\;lr\;hide"echo "aors=gvsip-$AccountName"echo "direct_media=no"echo "ice_support=yes"echo "rtcp_mux=yes"echo "media_use_received_transport=yes"echo "outbound_registration=gvsip-$AccountName"echo ""echo ""echo ""echo "In the FreePBX GUI:"echo ""echo "Add a custom trunk and make the trunk name: GVSIP-$AccountName"echo "Set the Outbound CallerID to: $GVPhoneNumber"echo "In the Dial Number Manipulation Rules tab add the pattern: 1|NXXNXXXXXX and also optionally aaa+NXXNXXXXXX (replace aaa with the assumed area code for 7 digit calls)"echo "In Custom Settings add the Custom Dial String: PJSIP/\$OUTNUM\$@gvsip-$AccountName"echo "Click Submit."echo ""echo "Add an Inbound Route with the Description: Google Voice - $UserFirstName - $PrefixCity"echo "Set the DID Number to: $GVPhoneNumber"echo "Set Destination to the desired extension, ring group, IVR, etc."echo "In the Advanced tab, set Force Answer to YES if you don't want to use Google's Voicemail."echo "In the Other tab, optionally enable a CID Lookup Source, or Enable Superfecta Lookup and select a Superfecta Scheme (probably Default)."echo "Click Submit."echo ""echo "Don't forget to add or modify Outbound Routes to use the GVSIP-$AccountName trunk and click Submit after each."echo ""echo "Click the Apply Config button in FreePBX, wait for the page to reload, then go to the Asterisk CLI and enter core restart now"echo "(this will interrupt any calls in progress) or core restart when convenient (which will delay the restart until there are no active calls)"echo ""echo ""
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Random gvsip outbound failures
I currently have 2 gvsip trunks using Bag updated code, configured but have this issue with 1 as well. About 50% of the time if I have not called out for a while I will get a failure on out bound calls. It eventually starts working but I don't know why.
It will attempt to use each outbound trunk but they all error out with
Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") i
Any help would be great
X[2018-07-14 22:26:01] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:30] Set("SIP/7002-0000000b", "post_num=@gvsip2") in new stack[2018-07-14 22:26:01] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:31] GotoIf("SIP/7002-0000000b", "1?outnum:skipoutnum") in new stack[2018-07-14 22:26:01] VERBOSE[19801][C-0000000b] pbx_builtins.c: Goto (macro-dialout-trunk,s,32)[2018-07-14 22:26:01] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:32] Set("SIP/7002-0000000b", "the_num=15158756513") in new stack[2018-07-14 22:26:01] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:33] Dial("SIP/7002-0000000b", "PJSIP/15158756513@gvsip2,300,T") in new stack[2018-07-14 22:26:01] VERBOSE[19801][C-0000000b] app_dial.c: Called PJSIP/15158756513@gvsip2[2018-07-14 22:26:01] DEBUG[19524] res_pjsip_outbound_registration.c: Found matching outbound registration state[2018-07-14 22:26:01] DEBUG[19524] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:AAYAYVANEUZDQ2EZE4CHEXCKL4FLV3ZOQXVKMSEXMCZVZUVP5KI7N5OCEVJLLZV:5060;uri-econt=X7AIJKY4V5EIXQESFN2GP3MEXR3MVSZG7O5JXBBD7EWCFP5HHUOFU536X7HK437PQDDTMTA7OJI4FYLRCHMDERPAIHAMUULMJUG4YVLD7WI7I5KKRK7OET2TJISQE22YSVZBIU;lr>[2018-07-14 22:26:01] DEBUG[19524] res_pjsip_outbound_registration.c: Found service-route. Adding route header for <sip:AA65NWTRGODM4RRZYUHJUPLWDEZNQFVENVJGYHFQQLRMCM2G6534GF2DMNGZ4YG:5060;transport=udp;lr;uri-econt=A5WAE4W23>[2018-07-14 22:26:01] ERROR[19524] pjproject: sip_msg Header with no vptr encountered!! Current buffer: INVITE sip:15158756513@obihai.sip.google.com SIP/2.0Via: SIP/2.0/TLS 173.19.138.122:5061;rport;branch=z9hG4bKPj2c026264-14ff-4925-aec7-94bbd7fa95e2;aliasFrom: <sip:5153443249@localhost.localdomain>;tag=8fedf826-7df8-401a-95b6-6c461dd4987aTo: <sip:15158756513@obihai.sip.google.com>Contact: <sip:asterisk@173.19.138.122:5061;transport=TLS>Call-ID: 8585e286-59ca-4be4-9d83-7757d7225d53CSeq: 29211 INVITE [2018-07-14 22:26:02] ERROR[19524] pjproject: sip_msg ..Header with no vptr encountered!! Current buffer: ACK sip:15158756513@obihai.sip.google.com SIP/2.0Via: SIP/2.0/TLS 173.19.138.122:5061;rport;branch=z9hG4bKPj2c026264-14ff-4925-aec7-94bbd7fa95e2;aliasFrom: <sip:5153443249@localhost.localdomain>;tag=8fedf826-7df8-401a-95b6-6c461dd4987aTo: <sip:15158756513@obihai.sip.google.com>;tag=480d0b23Call-ID: 8585e286-59ca-4be4-9d83-7757d7225d53CSeq: 29211 ACK [2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:34] NoOp("SIP/7002-0000000b", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:35] GotoIf("SIP/7002-0000000b", "1?continue,1:s-CHANUNAVAIL,1") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx_builtins.c: Goto (macro-dialout-trunk,continue,1)[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [continue@macro-dialout-trunk:1] NoOp("SIP/7002-0000000b", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/7002-0000000b", "1?Set(CALLERID(number)=7002)") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [5158756513@from-internal:6] Macro("SIP/7002-0000000b", "dialout-trunk,2,15158756513,,on") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:1] Set("SIP/7002-0000000b", "DIAL_TRUNK=2") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:2] ExecIf("SIP/7002-0000000b", "0?Set(DIAL_OPTIONS=tr)") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:3] GosubIf("SIP/7002-0000000b", "0?sub-pincheck,s,1()") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:4] ExecIf("SIP/7002-0000000b", "0?Set(CALLERID(num)=7002)") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:5] GotoIf("SIP/7002-0000000b", "0?disabletrunk,1") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:6] Set("SIP/7002-0000000b", "DIAL_NUMBER=15158756513") in new stack[2018-07-14 22:26:02] VERBOSE[19801][C-0000000b] pbx.c: Executing [s@macro-dialout-trunk:7] Set("SIP/7002-0000000b", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
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ObiHAI Obi100/Obi110 Firmware Mod Discussion
So many of us have the Obi1xx series devices that recently stopped connecting to google servers due to a certificate update. This thread is intended to discuss the possibility of modifying the firmware to update the certificate and let these devices work with Google Voice again.
These devices are based on a MIPS-X processor similar to the Sipura ATAs and there is not a lot of tools/docs out there about them except for a Yahoo Group mostly related to DVD player chipsets. The venerable DogFace05 who was an expert with these types of devices once posted that he was able to extract this firmware sucessfully. Not sure if he is still around. Anyone else familiar with this architecture?
It seems that the place to start looking is the end of the firmware update file which contains some kind of table. Then there seems to be a loader section which presumably decompresses one or more other sections and loads them to RAM before executing the firmware.
So the questions are:
Can we extract, modify, and repack the firmware and create proper checksums/signatures?
Where is the certificate stored and in what format?
Can we drop in a new certificate without messing up other things (e.g. if the length of the certificate has changed) or do we need to move the certificate and patch the code pointing to it?
Is updating the certificate enough or is the codebase missing support that is necessary (e.g. if key length has changed)?
Anyone who wants to participate please post your thoughts.
Thanks
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Localphone - No More Free iNums
Just received the below missive from Localphone (emphasis added):
Dear __________,
Localphone have been proud to be able to associated with Inums for the past 10 years, we have over that time provided the service free & absorbed all the cost into the business. However like everything things move on & changes have to be made.
It was deep regards that we have to announce that from July 1st 2018 we will be charging a fee to setup the Inums, the cost will be USD 0.25 with a Monthly charge of USD 0.25
Existing customers who have Inums on their account will not be charged a setup fee, but will be charged USD 0.25 per month from July 1st 2018.
Regards
Alan Smith
Operations Manager
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[Asterisk] PJSIP in Asterisk is hot stuff if you configure it right
I just wanted to share something that's not quite relevant to the GVSIP thread, but I have a very small "family and friends" system running Asterisk and FreePBX that I've been using as a gateway to Google Voice, among other things. With the ongoing demise of XMPP connectivity and a couple of other fortuitous events, I decided it was time to build a new system and of course I wanted to include naf's GVSIP packages. But I figured that if I had to use PJSIP anyway, I might as well see if I could get it to work for my extensions and trunks. I had a very low opinion of PJSIP going in and my initial tests were not encouraging; I had lots of issues. But to make a long story short, the thing that helped the most was adding this line as the very first line in /etc/asterisk/pjsip_custom_post.conf:
endpoint_identifier_order=auth_username,username,ip
And also, as I have described in another post, I found that setting tls - 0.0.0.0 - All to YES in the Chan_PJSIP settings caused problems with my test extension, but setting that to NO and then using naf's context:
[transport_tls]type=transportprotocol=tlsbind=0.0.0.0:5061
eliminated the issues, though I have no idea why, and I have no idea why others can use the FreePBX settings and have them work with no issues. But here is the thing that impressed me. I set all the extensions to use PJSIP, but I figured I was going to have a world of problems because many of them were connecting on ports other than 5060. In some cases it was a two line device connecting on ports 5060 and 5061; in other cases it was to avoid problems with routers or ISP's that seemed to think that traffic on port 5060 should be messed with. So my approach was going to be, bring up the new system and see what failed to connect, and either switch that extension back to Chan_SIP or if possible, try changing the device back to using port 5060. But you can imagine my shock and wonder when EVERY extension apparently connected without issues. And if I run pjsip show endpoints in the Asterisk CLI, the Contact: field shows the port each device is using. More wondrous is that the connections on port 5061 don't seem to interfere with the TLS stuff.
Maybe I have a fundamental misunderstanding of how SIP is supposed to work, and therefore I should not have been surprised, but it was never that easy in Chan_SIP. It seems to me that the biggest problem with PJSIP is the default configurations. For example, the default of IP matching first caused me a LOT of problems.
I only wanted to post this to say that maybe PJSIP is not something we should all avoid as long as possible. Sure, FreePBX needs to support it a bit better (how do I add the new keep_alive_interval=90 in the [global] section, per naf's instructions?) but I suspect they are learning too. I'm just happy as a clam that I converted all the extensions and trunks to PJSIP, and if you'd asked me a month ago I would not have had a good word to say about it!
Attached are my current General SIP settings (with IP address redacted, of course) and Chan PJSIP Settings pages. I'm not saying those are the optimum settings (comments are welcome) but they are part of what got it working for me.
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CircleNet has returned.... queue the ominous music
Hello DSLReports team,
I'm the owner of CircleNet LLC and previously was active on dslreports. I've been away from the board here and development on new CircleNet features has slowed to a crawl. But now we're back :-)
Why were you gone?
I've been taking care of a relative of ours that's suffered a brain aneurysm/stroke . It's been an uphill fight but she's 90% recovered now. SOOO much thanks to the patient folks at UVA.
Is stuff still running?
Yep, I've been care and feeding CircleNet.
So your back what's new?
Well we're now taking customers from the US (Sorry still no Ca).
Our porting cost has dropped to $10
Our interface has improved and FINALLY we have voice mail.
Our URL http://www.circlenet.biz/
What's the same?
Our call quality.
The awfulness of our website (Yep it's terrible).
Our great rates (Check out our simulator).
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[Future9] F9 Messed up again?
If I call someone F9 is sending a random (to me) caller ID.
If I try and call my ATT cell phone, which is working fine, F9 can't complete the call. If I use my COMCAST VOIP number to my cell phone it works fine.
This is terrible.
Nitzan, what is going on?
Thanks.
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[Future9] Future 9 down?
I can call out but those who call me, including myself from my cell, get a busy signal. Going to contact F9 myself now.
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FreePBX/Incredible PBX problems
So I have a small Incredible PBX OpenVZ VPS which is very quirky and I can never seem to figure out whats going on.
I have 1 DID from Callcentric and a few Google Voice DID's.
Here is the issue, Few calls will work for the CallCentric DID (Most important) Then randomly Callcentric will LOOSE connection with the VPS and incoming calls will give the callcentric generic error message. During this time the GUI of FreePBX is also inaccessible! BUT my Google Voice incoming lines work perfectly fine during this time. Give it 5 min and Callcentric and GUI will work again for a little and then stop again.
Its just odd that during the time callcentric DID stops working so does the GUI but the google voice incoming stuff still works fine. Server load on my end is low and using under 100mb of ram so resources I don't think are the issue.
I'm not sure if this is related to running it in OpenVZ? Or what?
Thanks!
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Anybody still on Teleblend?
It’s been 11 years almost to date since Sunrocket went down.. Is anyone still on Teleblend?
I noticed their webpage hasn’t been updated since 2015, and attempting to sign up gives an error.
Are they even still in business?
I wasn’t on this site back then, but I found myself without phone service and had to wait over 25 days to port back to BellSouth POTS and AMEX was able to get my $199 back thankfully.
I’ve kept a POTS line ever since the SunRocket fiasco.. Just never trusted another VoIP company since then.
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Testing your SIP TLS setup
I have been using a tool called testssl (https://github.com/drwetter/testssl.sh) for some time to test my SIP TLS setups. This is really useful because web-based SSL testers like ssllabs.com won't test your SIP port 5061.
Use it to test your own server: ./testssl.sh server.domain.name:5061
Here is a test of obihai.telephony.goog:5061: https://pastebin.com/3BXEjyBj I don't recommend scanning other people's services, but I think that if I am using a service regularly, then one scan is warranted.
I find it most useful in verifying the validity of the cert, including the chain (getting the intermediates in the right order), and making sure I have the right TLS versions enabled. It will show you a lot of output about ciphers. Some things you will not be able to do anything about depending on your software.
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[Voip.ms] VoIP.ms: New SMS Feature?
In VoIP.ms DID setup screen, in the SMS section, I just noticed a new option, "SMS SIP Account". Does anybody know what this function would be used for?
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What does Asterisk with obi200 give you direct connection to GV?
Hi all, I am trying to learn what options are out there. and I noticed some people using asterisk with their Obi200. What does Asterisk do for them over just connection from Obi200 straight to google voice?
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[Anveo] Anveo Direct inbound CNAM on Polycom: CID where CNAM should be
I just enabled CNAM lookups on my new Anveo Direct DID and I'm having a problem.
The endpoints (Polycom SoundPoint 450, 550) are displaying the CID where the CNAM should be.
The relevant sanitized INVITE header fields presented to the endpoint are:
XFrom: "COMPANY NAME" <sip:12125551234@50.22.101.14>;tag=7bcd63918ca620b9a0f03e44c02c4bec...P-Asserted-Identity: "12125551234" <sip:+12125551234@67.231.1.112:5060>
I'm not sure who is adding the P-Asserted-Identity header, but it's not present when I use the same ITSP (CallWithUs) to call my Callcentric DID, which routes through the same PBX to another line on the same endpoint.
The endpoints are behaving the way they are supposed to with voIpProt.SIP.CID.sourcePreference=Null (the default value): P-A-I has precedence over From.
I believe I could work around the problem with:
voIpProt.SIP.CID.sourcePreference=From,P-Asserted-Identity
on the endpoint but I haven't tried it yet.
My question is:
Is the P-Asserted-Identity header malformed because it has the CID where the display name should be?
An example INVITE from RFC 3325 has the following two P-A-I headers:
YP-Asserted-Identity: "Cullen Jennings" <sip:fluffy@cisco.com>P-Asserted-Identity: tel:+14085264000Thanks!
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Any real alternatives to the Obixxx devices for free home phone?
My Obi100 is still chugging along but I know it's only a matter of time before it stops working. I am looking into the Obi200 but haven't seen a sale on it in a long time. Are there any real alternatives to these devices that will give me a free home phone that I can connect a physical phone to? If not, I'll just have to get the obi200 I guess but just wanted to see what anyone else is doing.
Thanks!
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The Google Voice MOTIF Outbound Calling Stopped Working Thread
Thought I'd give this subject it's own thread to help it get solved. Mods please fee free to move the posts about this over to this thread.
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pygooglevoice --> oauth2
As much fun as it is hacking at yate, asterisk, and other VoIP software, I'm changing gears today and proposing a collaborative effort on an update of the old pygooglevoice script. Or rather, a total rewrite using current, robust API libraries and oauth2.
Direct SIP with Google is great but we are missing MWI and SMS support. It also seems like it would be good to get away from password auth because it's so touchy with Google.
Anyway, any interest? I'm not much of a programmer but can lend some insight. The API seems quite discoverable by using the developer console of your browser while using the GV web site. That, combined with some general knowledge of how to work with Google APIs (anyone done any work with GMail, etc?) seems like 50-75% of the task.
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[General] Running service no home server vs voip.ms
Hi everyone, this is more of a technical inquiry of running your own phone service, if this is even possible or worth it.
I currently have voip.ms service running on a PAP2T. It works great and performs flawlessly, but now i find myself wanted to do more with my home server, mostly just to tinker and mess around. I was thinking about cutting out voip.ms and run my own on said server.
Is this even possible? i understand that are some point, some monies has to be spent which i don't mind, but if it is possible, how would one achieve this? what are the pros and cons of this? I have seen some guides online but mostly pointed at using GV or some other free services. I don't mind continuing to pay voip.ms if needed for DID or other. Is it even worth it? Would i gain anything else that i currently do not get with voip.ms handling everything?
Thanks!
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